Computer-based multifunction personal communications system

ABSTRACT

A personal communications system is described which includes components of software and hardware operating in conjunction with a personal computer. The user interface control software operates on a personal computer, preferably within the Microsoft Windows® environment. The software control system communicates with hardware components linked to the software through the personal computer serial communications port. The hardware components include telephone communication equipment, digital signal processors, and hardware to enable voice, fax and data communication with a remote site connected through a standard telephone line. The functions of the hardware components are controlled by control software operating within the hardware component and from the software components operating within the personal computer. The major functions of the system are a telephone function, a voice mail function, a fax manager function, a multi-media mail function, a show and tell function, a terminal function and an address book function. The telephone function allows the present system to operate, from the users perspective, as a conventional telephone using either hands-free, headset or handset operation. The telephone function is more sophisticated than a standard telephone in that the present system converts the voice into a digital signal which can be processed with echo cancellation, compressed, stored as digital data for later retrieval and transmitted as digital voice data concurrent with the transfer of digital information data.

This is a division of application Ser. No. 08/002,467, filed Jan. 8,1993. Microfiche appendix is included in the application.

FIELD OF THE INVENTION

The present invention relates to communications systems and inparticular to computer assisted digital communications including data,fax and digitized voice.

BACKGROUND OF THE INVENTION

A wide variety of communications alternatives are currently available totelecommunications users. For example, facsimile transmission of printedmatter is available through what is commonly referred to as astand-alone fax machine. Alternatively, fax-modem communication systemsare currently available for personal computer users which combine theoperation of a facsimile machine with the word processor of a computerto transmit documents held on computer disk. Modem communication overtelephone lines in combination with a personal computer is also known inthe art where file transfers can be accomplished from one computer toanother. Also, simultaneous voice and modem data transmitted over thesame telephone line has been accomplished in several ways.

There is a need in the art, however, for a personal communicationssystem which combines a wide variety of communication functions into anintegrated hardware-software product such that the user can convenientlychoose a mode of communication and have that communication automaticallyinvoked from a menu driven selection system.

SUMMARY OF THE INVENTION

The present disclosure describes a complex computer assistedcommunications system which contains multiple inventions. The subject ofthe present multiple inventions is a personal communications systemwhich includes components of software and hardware operating inconjunction with a personal computer. The user interface controlsoftware operates on a personal computer, preferably within theMicrosoft Windows® environment. The software control system communicateswith hardware components linked to the software through the personalcomputer serial communications port. The hardware components includetelephone communication equipment, digital signal processors, andhardware to enable both fax and data communication with a hardwarecomponents at a remote site connected through a standard telephone line.The functions of the hardware components are controlled by controlsoftware operating within the hardware component and from the softwarecomponents operating within the personal computer.

Communications between the software components running on the personalcomputer and the local hardware components over the serialcommunications link is by a special packet protocol for digital datacommunications. This bi-directional communications protocol allowsuninterrupted bi-directional full-duplex transfer of both controlinformation and data communication.

The major functions of the present system are a telephone function, avoice mail function, a fax manager function, a multi-media mailfunction, a show and tell function, a terminal function and an addressbook function. The telephone function allows the present system tooperate, from the users perspective, as a conventional telephone usingeither hands-free, headset or handset operation. The telephone functionis more sophisticated than a standard telephone in that the presentsystem converts the voice into a digital signal which can be processedwith echo cancellation, compressed, stored as digital data for laterretrieval and transmitted as digital voice data concurrent with thetransfer of digital information data.

The voice mail function of the present system operates as a telephoneanswering machine which can receive, compress and store voice messagesfor later retrieval or reuse in response messaging.

The fax manager function of the present system allows the transmissionand reception of facsimile information. The software component of thepresent system operates in conjunction with other commercially availablesoftware programs such as word processors and the like to transmit andreceive facsimile pages of digital data stored on a computer system.

The multi-media mail component of the present system allows the operatorto create documents that include text, graphics and voice mail messageswhich can be sent as a combined package over conventional telephonelines for receipt at a like-configured site using the present system.

The show and tell component of the present system enables the operatorto simultaneously transmit voice and data communication to a remotesite. This voice over data function dynamically allocates data bandwidthover the telephone line depending on the demands of the voice gradedigitized signal.

The terminal feature of the present system allows the user to establisha data communications session with another computer system allowing theuser's local computer system to operate as a dumb terminal.

The address book function of the present system is a versatile databasethat is built by the user and operates in conjunction with the othercomponents of the present system to dial and establish communicationlinks with remote sites to enable data communication, voice mail,facsimile, file transfer all in an automated mode without userintervention.

The hardware components of the present system include circuitry toenable digital data communication and facsimile communication overstandard telephone lines. The hardware components also include circuitryto convert the voice to digital data and compress that data for transferto the software component on the personal computer or transfer it overthe telephone lines to a remote site.

Many of the functions of the present system are accomplished byincluding a voice control digital signal processor (DSP) to operate inconjunction with a data/fax modem implemented with a data pump DSP. Thedata pump DSP and the voice control DSP accomplish the followingfunctions in an integrated hardware arrangement:

A sophisticated telephone apparatus with its attached handset, headsetand a built-in hands free telephone operation using the integratedmicrophone and speaker system. The hands free telephone works in fullduplex mode through the use of voice echo cancellation performed by thevoice control DSP.

The voice control DSP in conjunction with a telephone CODEC providesvoice compression which can be sent to the computer system that isattached the RS232 port for storage and later retrieval. The compressedvoice from the voice control DSP can also be multiplexed with the inputdata stream from the personal computer with dynamic time allocation.Whereas, the input data from the attached computer is transmitted usingthe error control protocol like MNP or V.42 with or without datacompression (e.g., V.42 bis), the speech is packetized using a differentheader defining it as a speech packet and then transmitted through acontroller. The speech packets, like the data packets, have the attachedCRC codes. However, the speech packets are not sequenced and the likehardware at the receiving end ignores the accompanying CRC codes forvoice packets and passes the voice packets to the voice control DSP fordecompression. The decompressed speech is played through one of thetelephone receiving units, i.e., the headset, handset or the built inspeaker.

The voice control DSP allows the compressed speech to be recorded on arecording media, e.g., the hard disc drive of the attached computersystem. This provides the function of an answering machine. In additionto the answering machine function, the recorded speech can be providedfor the voice mail functions.

The special packet protocol over the RS232 interface between thesoftware component and the hardware component that governs the operationof the hardware component is so designed that it allows various controlfunctions to be intermixed with data over the RS232 serial port. Thesoftware component of the present system accepts the generic AT modemcommands when not in the special packet mode. When the hardwarecomponent is configured to accept the packet level protocol over theRS232 port, it can be made to switch to the generic command mode throughthe use of a break sequence.

The hardware components of the present system functions as a data/faxmodem when the speech compression or telephone mode is not invoked. Thepacket mode or the generic AT command mode may be used for this purpose.

The hardware components of the present system incorporate a provisionfor a special link integrity packet to facilitate the device to workover cellular networks. This scheme allows the modem in one of itsplurality of modes to ignore the carrier dropouts (selective fading)inherent in the cellular networks. Such a scheme does not use carrierdetect circuitry of the modem. The disconnect of the cellular connectionis done through a negotiation scheme using packet interchange betweenthe two ends of the link.

In cellular networks the multiplexed voice data technology of thepresent system allows a single apparatus to function as a smarttelephone, an intelligent data modem as well as a fax modem. Thesefeatures along with the voice data multiplex mode provides a travelinguser complete freedom to use his or her moving vehicle as a truetraveling office.

These features of the hardware component of the present system alongwith the features of the software component of the present systemrunning on a PC provides a user with a complete range oftelecommunications functions of a modern office, be it a stationary ormobile.

DESCRIPTION OF THE DRAWINGS

In the drawings, where like numerals describe like components throughoutthe several views,

FIG. 1 shows the telecommunications environment within which the presentmay operate in several of the possible modes of communication;

FIG. 2 is the main menu icon for the software components operating onthe personal computer;

FIG. 3 is a block diagram of the hardware components of the presentsystem;

FIG. 4 is a key for viewing the detailed electrical schematic diagramsof FIGS. 5A-10C to facilitate understanding of the interconnect betweenthe drawings;

FIGS. 5A-5C, 6A-6C, 7A-7C, 8A-8B, 9A-9C and 10A-10C are detailedelectrical schematic diagrams of the circuitry of the hardwarecomponents of the present system;

FIG. 11 is a signal flow diagram of the speech compression algorithm;

FIG. 12 is a detailed function flow diagram of the speech compressionalgorithm;

FIG. 13 is a detailed function flow diagram of the speech decompressionalgorithm;

FIG. 14 is a detailed function flow diagram of the echo cancellationalgorithm;

FIG. 15 is a detailed function flow diagram of the voice/datamultiplexing function;

FIG. 16 is a perspective view of the components of a digital computercompatible with the present invention;

FIG. 17 is a block diagram of the software structure compatible with thepresent invention;

FIG. 18 is a block diagram of the control structure of softwarecompatible with the present invention;

FIG. 19 is a block diagram of the main menu structure of softwarecompatible with the present invention;

FIG. 20 is a flow diagram of answer mode software compatible with thepresent invention;

FIG. 21 is a flow diagram of telephone software compatible with thepresent invention;

FIG. 22 is a flow diagram of voice mail software compatible with thepresent invention;

FIG. 23 is a flow diagram of fax manager software compatible with thepresent invention;

FIG. 24 is a flow diagram of multi-media mail software compatible withthe present invention;

FIG. 25 is a flow diagram of a timing loop compatible with the presentinvention;

FIG. 26 is a flow diagram of telephone control software compatible withthe present invention;

FIG. 27 is a flow diagram of voice mail control software compatible withthe present invention;

FIG. 28 is a flow diagram of high resolution fax driver softwarecompatible with the present invention;

FIG. 29 is a flow diagram of low resolution fax driver softwarecompatible with the present invention;

FIG. 30 is a flow diagram of multi media mail control softwarecompatible with the present invention;

FIG. 31 is a flow diagram of multi media mail editor software compatiblewith the present invention;

FIG. 32 is a flow diagram of multi media mail transmit softwarecompatible with the present invention;

FIG. 33 is a flow diagram of multi media mail receive softwarecompatible with the present invention;

FIG. 34 is a flow diagram of show and tell transmit software compatiblewith the present invention;

FIG. 35 is a flow diagram of show and tell receive software compatiblewith the present invention;

FIG. 36 is a flow diagram of voice mail transmit software compatiblewith the present invention;

FIG. 37 is a flow diagram of voice mail receive software compatible withthe present invention;

FIG. 38 is a flow diagram of an outgoing timer loop compatible with thepresent invention;

FIG. 39 is a flow diagram of an outgoing timer loop compatible with thepresent invention;

FIG. 40 is an initialization screen display compatible with the presentinvention;

FIG. 41 is a communication port setup screen display compatible with thepresent invention;

FIG. 42 is an answer mode setup screen display compatible with thepresent invention;

FIG. 43 is a hold call setup screen display compatible with the presentinvention;

FIG. 44 is a voice mail setup screen display compatible with the presentinvention;

FIG. 45 is a PBX setup screen display compatible with the presentinvention;

FIG. 46 is a fax setup screen display compatible with the presentinvention;

FIG. 47 is a multi-media mail setup screen display compatible with thepresent invention;

FIG. 48 is a show and tell setup screen display compatible with thepresent invention;

FIG. 49 is a telephone control screen display compatible with thepresent invention;

FIG. 50 is a voice mail control screen display compatible with thepresent invention;

FIG. 51 is a voice editor screen display compatible with the presentinvention;

FIG. 52 is a fax manager control screen display compatible with thepresent invention;

FIG. 53 is a multi-media mail control screen display compatible with thepresent invention;

FIG. 54 is a show and tell control screen display compatible with thepresent invention;

FIG. 55 is an address book control screen display compatible with thepresent invention;

FIG. 56 is a voice message destination screen display compatible withthe present invention; and

FIG. 57 is a message composer screen display compatible with the presentinvention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

The specification for the multiple inventions described herein includesthe present description, the drawings and a microfiche appendix. In thefollowing detailed description of the preferred embodiment, reference ismade to the accompanying drawings which form a part hereof, and in whichis shown by way of illustration specific embodiments in which theinventions may be practiced. These embodiments are described insufficient detail to enable those skilled in the art to practice theinvention, and it is to be understood that other embodiments may beutilized and that structural changes may be made without departing fromthe spirit and scope of the present inventions. The following detaileddescription is, therefore, not to be taken in a limiting sense, and thescope of the present inventions is defined by the appended claims.

FIG. 1 shows a typical arrangement for the use of the present system.Personal computer 10 is running the software components of the presentsystem while the hardware components 20 include the data communicationequipment and telephone headset. Hardware components 20 communicate overa standard telephone line 30 to one of a variety of remote sites. One ofthe remote sites may be equipped with the present system includinghardware components 20a and software components running on personalcomputer 10a. In one alternative use, the local hardware components 20may be communicating over standard telephone line 30 to facsimilemachine 60. In another alternative use, the present system may becommunicating over a standard telephone line 30 to another personalcomputer 80 through a remote modem 70. In another alternative use, thepresent system may be communicating over a standard telephone line 30 toa standard telephone 90. Those skilled in the art will readily recognizethe wide variety of communication interconnections possible with thepresent system by reading and understanding the following detaileddescription.

The ornamental features of the hardware components 20 of FIG. 1 areclaimed as part of Design patent application Number 29/001368, filedNov. 12, 1992 entitled "Telephone/Modem case for a Computer-BasedMultifunction Personal Communications System" assigned to the sameassignee of the present inventions and hereby incorporated by reference.

General Overview

The present inventions are embodied in a commercial product by theassignee, MultiTech Systems, Inc. The software component operating on apersonal computer is sold under the commercial trademark ofMultiExpressPCS™ personal communications software while the hardwarecomponent of the present system is sold under the commercial name ofMultiModemPCS™, Intelligent Personal Communications System Modem. In thepreferred embodiment, the software component runs under Microsoft®Windows® however those skilled in the art will readily recognize thatthe present system is easily adaptable to run under any single ormulti-user, single or multi-window operating system.

The present system is a multifunction communication system whichincludes hardware and software components. The system allows the user toconnect to remote locations equipped with a similar system or withmodems, facsimile machines or standard telephones over a single analogtelephone line. The software component of the present system includes anumber of modules which are described in more detail below.

FIG. 2 is an example of the Windows®-based main menu icon of the presentsystem operating on a personal computer. The functions listed with theicons used to invoke those functions are shown in the preferredembodiment. Those skilled in the art will readily recognize that a widevariety of selection techniques may be used to invoke the variousfunctions of the present system. The icon of FIG. 2 is part of Designpatent application Number 29/001397, filed Nov. 12, 1992 entitled "Iconsfor a Computer-Based Multifunction Personal Communications System"assigned to the same assignee of the present inventions and herebyincorporated by reference.

The telephone module allows the system to operate as a conventional orsophisticated telephone system. The system converts voice into a digitalsignal so that it can be transmitted or stored with other digital data,like computer information. The telephone function supports PBX andCentrex features such a call waiting, call forwarding, caller ID andthree-way calling. This module also allows the user to mute, hold orrecord a conversation. The telephone module enables the handset, headsetor hands-free speaker telephone operation of the hardware component. Itincludes on-screen push button dialing, speed-dial of stored numbers anddigital recording of two-way conversations.

The voice mail portion of the present system allows this system tooperate as a telephone answering machine by storing voice messages asdigitized voice files along with a time/date voice stamp. The digitizedvoice files can be saved and sent to one or more destinationsimmediately or at a later time using a queue scheduler. The user canalso listen to, forward or edit the voice messages which have beenreceived with a powerful digital voice editing component of the presentsystem. This module also creates queues for outgoing messages to be sentat preselected times and allows the users to create outgoing messageswith the voice editor.

The fax manager portion of the present system is a queue for incomingand outgoing facsimile pages. In the preferred embodiment of the presentsystem, this function is tied into the Windows "print" command once thepresent system has been installed. This feature allows the user tocreate faxes from any Windows®-based document that uses the "print"command. The fax manager function of the present system allows the userto view queued faxes which are to be sent or which have been received.This module creates queues for outgoing faxes to be sent at preselectedtimes and logs incoming faxes with time/date stamps.

The multi-media mail function of the present system is a utility whichallows the user to compose documents that include text, graphics andvoice messages using the message composer function of the presentsystem, described more fully below. The multi-media mail utility of thepresent system allows the user to schedule messages for transmittal andqueues up the messages that have been received so that can be viewed ata later time.

The show and tell function of the present system allows the user toestablish a data over voice (DOV) communications session. When the useris transmitting data to a remote location similarly equipped, the useris able to talk to the person over the telephone line while concurrentlytransferring the data. This voice over data function is accomplished inthe hardware components of the present system. It digitizes the voiceand transmits it in a dynamically changing allocation of voice data anddigital data multiplexed in the same transmission. The allocation at agiven moment is selected depending on the amount of voice digitalinformation required to be transferred. Quiet voice intervals allocategreater space to the digital data transmission.

The terminal function of the present system allows the user to establisha data communications session with another computer which is equippedwith a modem but which is not equipped with the present system. Thisfeature of the present system is a Windows®-based data communicationsprogram that reduces the need for issuing "AT" commands by providingmenu driven and "pop-up" window alternatives.

The address book function of the present system is a database that isaccessible from all the other functions of the present system. Thisdatabase is created by the user inputting destination addresses andtelephone numbers for data communication, voice mail, facsimiletransmission, modem communication and the like. The address bookfunction of the present system may be utilized to broadcastcommunications to a wide variety of recipients. Multiple linkeddatabases have separate address books for different groups and differentdestinations may be created by the users. The address book functionincludes a textual search capability which allows fast and efficientlocation of specific addresses as described more fully below.

Hardware Components

FIG. 3 is a block diagram of the hardware components of the presentsystem corresponding to reference number 20 of FIG. 1. These componentsform the link between the user, the personal computer running thesoftware component of the present system and the telephone lineinterface. As will be more fully described below, the interface to thehardware components of the present system is via a serial communicationsport connected to the personal computer. The interface protocol is wellordered and defined such that other software systems or programs runningon the personal computer may be designed and implemented which would becapable of controlling the hardware components shown in FIG. 3 by usingthe control and communications protocol defined below.

In the preferred embodiment of the present system three alternatetelephone interfaces are available: the telephone handset 301, atelephone headset 302, and a hands-free microphone 303 and speaker 304.Regardless of the telephone interface, the three alternative interfacesconnect to the digital telephone coder-decoder (CODEC) circuit 305.

The digital telephone CODEC circuit 305 interfaces with the voicecontrol digital signal processor (DSP) circuit 306 which includes avoice control DSP and CODEC. This circuit does digital to analog (D/A)conversion, analog to digital (A/D) conversion, coding/decoding, gaincontrol and is the interface between the voice control DSP circuit 306and the telephone interface. The CODEC of the voice control circuit 306transfers digitized voice information in a compressed format tomultiplexor circuit 310 to analog telephone line interface 309.

The CODEC of the voice control circuit 306 is actually an integralcomponent of a voice control digital signal processor integratedcircuit, as described more fully below. The voice control DSP of circuit306 controls the digital telephone CODEC circuit 305, performs voicecompression and echo cancellation.

Multiplexor (MUX) circuit 310 selects between the voice control DSPcircuit 306 and the data pump DSP circuit 311 for transmission ofinformation on the telephone line through telephone line interfacecircuit 309.

The data pump circuit 311 also includes a digital signal processor (DSP)and a CODEC for communicating over the telephone line interface 309through MUX circuit 310. The data pump DSP and CODEC of circuit 311performs functions such as modulation, demodulation and echocancellation to communicate over the telephone line interface 309 usinga plurality of telecommunications standards including FAX and modemprotocols.

The main controller circuit 313 controls the DSP data pump circuit 311and the voice control DSP circuit 306 through serial input/output andclock timer control (SIO/CTC) circuits 312 and dual port RAM circuit 308respectively. The main controller circuit 313 communicates with thevoice control DSP 306 through dual port RAM circuit 308. In this fashiondigital voice data can be read and written simultaneously to the memoryportions of circuit 308 for high speed communication between the user(through interfaces 301, 302 or 303/304) and the personal computerconnected to serial interface circuit 315 and the remote telephoneconnection connected through the telephone line attached to lineinterface circuit 309.

As described more fully below, the main controller circuit 313 includes,in the preferred embodiment, a microprocessor which controls thefunctions and operation of all of the hardware components shown in FIG.3. The main controller is connected to RAM circuit 316 and anprogrammable and electrically erasable read only memory (PEROM) circuit317. The PEROM circuit 317 includes nonvolatile memory in which theexecutable control programs for the voice control DSP circuits 306 andthe main controller circuits 313 operate.

The RS232 serial interface circuit 315 communicates to the serial portof the personal computer which is running the software components of thepresent system. The RS232 serial interface circuit 315 is connected to aserial input/output circuit 314 with main controller circuit 313. SIOcircuit 314 is in the preferred embodiment, a part of SIO/CTC circuit312.

Functional Operation of the Hardware Components

Referring once again to FIG. 3, the multiple and selectable functionsdescribed in conjunction with FIG. 2 are all implemented in the hardwarecomponents of FIG. 3. Each of these functions will be discussed in turn.

The telephone function 115 is implemented by the user either selecting atelephone number to be dialed from the address book 127 or manuallyselecting the number through the telephone menu on the personalcomputer. The telephone number to be dialed is downloaded from thepersonal computer over the serial interface and received by maincontroller 313. Main controller 313 causes the data pump DSP circuit 311to seize the telephone line and transmit the DTMF tones to dial anumber. Main controller 313 configures digital telephone CODEC circuit305 to enable either the handset 301 operation, the microphone 303 andspeaker 304 operation or the headset 302 operation. A telephoneconnection is established through the telephone line interface circuit309 and communication is enabled. The user's analog voice is transmittedin an analog fashion to the digital telephone CODEC 305 where it isdigitized. The digitized voice patterns are passed to the voice controlcircuit 306 where echo cancellation is accomplished, the digital voicesignals are reconstructed into analog signals and passed throughmultiplexor circuit 310 to the telephone line interface circuit 309 foranalog transmission over the telephone line. The incoming analog voicefrom the telephone connection through telephone connection circuit 309is passed to the integral CODEC of the voice control circuit 306 whereit is digitized. The digitized incoming voice is then passed to digitaltelephone CODEC circuit 305 where it is reconverted to an analog signalfor transmission to the selected telephone interface (either the handset301, the microphone/speaker 303/304 or the headset 302). Voice ControlDSP circuit 306 is programmed to perform echo cancellation to avoidfeedback and echoes between transmitted and received signals, as is morefully described below.

In the voice mail function mode of the present system, voice messagesmay be stored for later transmission or the present system may operateas an answering machine receiving incoming messages. For storingdigitized voice, the telephone interface is used to send the analogspeech patterns to the digital telephone CODEC circuit 305. Circuit 305digitizes the voice patterns and passes them to voice control circuit306 where the digitized voice patterns are digitally compressed. Thedigitized and compressed voice patterns are passed through dual port ramcircuit 308 to the main controller circuit 313 where they aretransferred through the serial interface to the personal computer usinga packet protocol defined below. The voice patterns are then stored onthe disk of the personal computer for later use in multi-media mail, forvoice mail, as a pre-recorded answering machine message or for laterpredetermined transmission to other sites.

For the present system to operate as an answering machine, the hardwarecomponents of FIG. 3 are placed in answer mode. An incoming telephonering is detected through the telephone line interface circuit 309 andthe main controller circuit 313 is alerted which passes the informationoff to the personal computer through the RS232 serial interface circuit315. The telephone line interface circuit 309 seizes the telephone lineto make the telephone connection. A pre-recorded message may be sent bythe personal computer as compressed and digitized speech through theRS232 interface to the main controller circuit 313. The compressed anddigitized speech from the personal computer is passed from maincontroller circuit 313 through dual port ram circuit 308 to the voicecontrol DSP circuit 306 where it is uncompressed and converted to analogvoice patterns. These analog voice patterns are passed throughmultiplexor circuit 310 to the telephone line interface 309 fortransmission to the caller. Such a message may invite the caller toleave a voice message at the sound of a tone. The incoming voicemessages are received through telephone line interface 309 and passed tovoice control circuit 306. The analog voice patterns are digitized bythe integral CODEC of voice control circuit 306 and the digitized voicepatterns are compressed by the voice control DSP of the voice controlcircuit 306. The digitized and compressed speech patterns are passedthrough dual port ram circuit 308 to the main controller circuit 313where they are transferred using packet protocol described below throughthe RS232 serial interface 315 to the personal computer for storage andlater retrieval. In this fashion the hardware components of FIG. 3operate as a transmit and receive voice mail system for implementing thevoice mail function 117 of the present system.

The hardware components of FIG. 3 may also operate to facilitate the faxmanager function 119 of FIG. 2. In fax receive mode, an incomingtelephone call will be detected by a ring detect circuit of thetelephone line interface 309 which will alert the main controllercircuit 313 to the incoming call. Main controller circuit 313 will causeline interface circuit 309 to seize the telephone line to receive thecall. Main controller circuit 313 will also concurrently alert theoperating programs on the personal computer through the RS232 interfaceusing the packet protocol described below. Once the telephone lineinterface seizes the telephone line, a fax carrier tone is transmittedand a return tone and handshake is received from the telephone line anddetected by the data pump circuit 311. The reciprocal transmit andreceipt of the fax tones indicates the imminent receipt of a facsimiletransmission and the main controller circuit 313 configures the hardwarecomponents of FIG. 3 for the receipt of that information. The necessaryhandshaking with the remote facsimile machine is accomplished throughthe data pump 311 under control of the main controller circuit 313. Theincoming data packets of digital facsimile data are received over thetelephone line interface and passed through data pump circuit 311 tomain controller circuit 313 which forwards the information on a packetbasis (using the packet protocol described more fully below) through theserial interface circuit 315 to the personal computer for storage ondisk. Those skilled in the art will readily recognize that the FAX datacould be transferred from the telephone line to the personal computerusing the same path as the packet transfer except using the normal ATstream mode. Thus, the incoming facsimile is automatically received andstored on the personal computer through the hardware components of FIG.3.

A facsimile transmission is also facilitated by the hardware componentsof FIG. 3. The transmission of a facsimile may be immediate or queuedfor later transmission at a predetermined or preselected time. Controlpacket information to configure the hardware components to send afacsimile are sent over the RS232 serial interface between the personalcomputer and the hardware components of FIG. 3 and are received by maincontroller circuit 313. The data pump circuit 311 then dials therecipient's telephone number using DTMF tones or pulse dialing over thetelephone line interface circuit 309. Once an appropriate connection isestablished with the remote facsimile machine, standard facsimilehandshaking is accomplished by the data pump circuit 311. Once thefacsimile connection is established, the digital facsimile pictureinformation is received through the data packet protocol transfer overserial line interface circuit 315, passed through main controllercircuit 313 and data pump circuit 311 onto the telephone line throughtelephone line interface circuit 309 for receipt by the remote facsimilemachine.

The operation of the multi-media mail function 121 of FIG. 2 is alsofacilitated by the hardware components of FIG. 3. A multimediatransmission consists of a combination of picture information, digitaldata and digitized voice information. For example, the type ofmultimedia information transferred to a remote site using the hardwarecomponents of FIG. 3 could be the multimedia format of the MicroSoft®Multimedia Wave® format with the aid of an Intelligent Serial Interface(ISI) card added to the personal computer. The multimedia may also bethe type of multimedia information assembled by the software componentof the present system-which is described more fully below.

The multimedia package of information including text, graphics and voicemessages (collectively called the multimedia document) may betransmitted or received through the hardware components shown in FIG. 3.For example, the transmission of a multimedia document through thehardware components of FIG. 3 is accomplished by transferring themultimedia digital information using the packet protocol described belowover the RS232 serial interface between the personal computer and theserial line interface circuit 315. The packets are then transferredthrough main controller circuit 313 through the data pump circuit 311 onto the telephone line for receipt at a remote site through telephoneline interface circuit 309. In a similar fashion, the multimediadocuments received over the telephone line from the remote site arereceived at the telephone line interface circuit 309, passed through thedata pump circuit 311 for receipt and forwarding by the main controllercircuit 313 over the serial line interface circuit 315.

The show and tell function 123 of the present system allows the user toestablish a data over voice communication session. In this mode ofoperation, full duplex data transmission may be accomplishedsimultaneously with the voice communication between both sites. Thismode of operation assumes a like configured remote site. The hardwarecomponents of the present system also include a means for sendingvoice/data over cellular links. The protocol used for transmittingmultiplexed voice and data include a supervisory packet described morefully below to keep the link established through the cellular link. Thissupervisory packet is an acknowledgement that the link is still up. Thesupervisory packet may also contain link information to be used foradjusting various link parameters when needed. This supervisory packetis sent every second when data is not being sent and if the packet isnot acknowledged after a specified number of attempts, the protocolwould then give an indication that the cellular link is down and thenallow the modem to take action. The action could be for example; changespeeds, retrain, or hang up. The use of supervisory packets is a novelmethod of maintaining inherently intermittent cellular links whentransmitting multiplexed voice and data.

The voice portion of the voice over data transmission of the show andtell function is accomplished by receiving the user's voice through thetelephone interface 301, 302 or 303 and the voice information isdigitized by the digital telephone circuit 305. The digitized voiceinformation is passed to the voice control circuit 306 where thedigitized voice information is compressed using a voice compressionalgorithm described more fully below. The digitized and compressed voiceinformation is passed through dual port RAM circuit 308 to the maincontroller circuit 313. During quiet periods of the speech, a quiet flagis passed from voice control circuit 306 to the main controller 313through a packet transfer protocol described below by a dual port RAMcircuit 308.

Simultaneous with the digitizing compression and packetizing of thevoice information is the receipt of the packetized digital informationfrom the personal computer over interface line circuit 315 by maincontroller circuit 313. Main controller circuit 313 in the show and tellfunction of the present system must efficiently and effectively combinethe digitized voice information with the digital information fortransmission over the telephone line via telephone line interfacecircuit 309. As described above and as described more fully below, maincontroller circuit 313 dynamically changes the amount of voiceinformation and digital information transmitted at any given period oftime depending upon the quiet times during the voice transmissions. Forexample, during a quiet moment where there is no speech informationbeing transmitted, main controller circuit 313 ensures that a highervolume of digital data information be transmitted over the telephoneline interface in lieu of digitized voice information.

Also, as described more fully below, the packets of digital datatransmitted over the telephone line interface with the transmissionpacket protocol described below, requires 100 percent accuracy in thetransmission of the digital data, but a lesser standard of accuracy forthe transmission and receipt of the digitized voice information. Sincedigital information must be transmitted with 100 percent accuracy, acorrupted packet of digital information received at the remote site mustbe re-transmitted. A retransmission signal is communicated back to thelocal site and the packet of digital information which was corruptedduring transmission is retransmitted. If the packet transmittedcontained voice data, however, the remote site uses the packets whetherthey were corrupted or not as long as the packet header was intact. Ifthe header is corrupted, the packet is discarded. Thus, the voiceinformation may be corrupted without requesting retransmission since itis understood that the voice information must be transmitted on a realtime basis and the corruption of any digital information of the voicesignal is not critical. In contrast to this the transmission of digitaldata is critical and retransmission of corrupted data packets isrequested by the remote site.

The transmission of the digital data follows the CCITT V.42 standard, asis well known in the industry and as described in the CCITT Blue Book,volume VIII entitled Data Communication over the Telephone Network,1989. The CCITT V.42 standard is hereby incorporated by reference. Thevoice data packet information also follows the CCITT V.42 standard, butuses a different header format so the receiving site recognizes thedifference between a data packet and a voice packet. The voice packet isdistinguished from a data packet by using undefined bits in the header(80 hex) of the V.42 standard. The packet protocol for voice over datatransmission during the show and tell function of the present system isdescribed more fully below.

Since the voice over data communication with the remote site isfull-duplex, incoming data packets and incoming voice packets arereceived by the hardware components of FIG. 3. The incoming data packetsand voice packets are received through the telephone line interfacecircuit 309 and passed to the main controller circuit 313 via data pumpDSP circuit 311. The incoming data packets are passed by the maincontroller circuit 313 to the serial interface circuit 315 to be passedto the personal computer. The incoming voice packets are passed by themain controller circuit 313 to the dual port RAM circuit 308 for receiptby the voice control DSP circuit 306. The voice packets are decoded andthe compressed digital information therein is uncompressed by the voicecontrol DSP of circuit 306. The uncompressed digital voice informationis passed to digital telephone CODEC circuit 305 where it is reconvertedto an analog signal and retransmitted through the telephone lineinterface circuits. In this fashion full-duplex voice and datatransmission and reception is accomplished through the hardwarecomponents of FIG. 3 during the show and tell functional operation ofthe present system.

Terminal operation 125 of the present system is also supported by thehardware components of FIG. 3. Terminal operation means that the localpersonal computer simply operates as a "dumb" terminal including filetransfer capabilities. Thus no local processing takes place other thanthe handshaking protocol required for the operation of a dumb terminal.In terminal mode operation, the remote site is assumed to be a modemconnected to a personal computer but the remote site is not necessarilya site which is configured according to the present system. In terminalmode of operation, the command and data information from personalcomputer is transferred over the RS232 serial interface circuit 315,forwarded by main controller circuit 313 to the data pump circuit 311where the data is placed on the telephone line via telephone lineinterface circuit 309.

In a reciprocal fashion, data is received from the telephone line overtelephone line interface circuit 309 and simply forwarded by the datapump circuit 311, the main controller circuit 313 over the serial lineinterface circuit 315 to the personal computer.

As described above, and more fully below, the address book function ofthe present system is primarily a support function for providingtelephone numbers and addresses for the other various functions of thepresent system.

Detailed Electrical Schematic Diagrams

The detailed electrical schematic diagrams comprise FIGS. 5A-C, 6A-C,7A-C, 8A-B, 9A-C and 10A-C. FIG. 4 shows a key on how the schematicdiagrams may be conveniently arranged to view the passing of signals onthe electrical lines between the diagrams. The electrical connectionsbetween the electrical schematic diagrams are through the designatorslisted next to each wire. For example, on the right side of FIG. 5A,address lines A0-A19 are attached to an address bus for which theindividual electrical lines may appear on other pages as A0-A19 or maycollectively be connected to other schematic diagrams through thedesignator "A" in the circle connected to the collective bus. In a likefashion, other electrical lines designated with symbols such as RNGL onthe lower left-hand side of FIG. 5A may connect to other schematicdiagrams using the same signal designator RNGL.

Beginning with the electrical schematic diagram of FIG. 7C, thetelephone line connection in the preferred embodiment is throughconnector J2 which is a standard six-pin modular RJ-11 Jack. In theschematic diagram of FIG. 7C, only the tip and ring connections of thefirst telephone circuit of the RJ-11 modular connector are used. Ferritebeads FB3 and FB4 are placed on the tip and ring wires of the telephoneline connections to remove any high frequency or RF noise on theincoming telephone line. The incoming telephone line is also overvoltageprotected through SIDACTOR R4. The incoming telephone line may be fullwave rectified by the full wave bridge comprised of diodes CR27, CR28,CR29 and CR31. Switch S4 switches between direct connection and fullwave rectified connection depending upon whether the line is anon-powered leased line or a standard telephone line. Since a leasedline is a "dead" line with no voltage, the full-wave rectification isnot needed.

Also connected across the incoming telephone line is a ring detectcircuit. Optical isolator U32 (part model number CNY17) senses the ringvoltage threshold when it exceeds the breakdown voltages on zener diodesCR1 and CR2. A filtering circuit shown in the upper right corner of FIG.7C creates a long RC delay to sense the constant presence of an AC ringvoltage and buffers that signal to be a binary signal out of operationalamplifier U25 (part model number TLO82). Thus, the RNGL and J1RINGsignals are binary signals for use in the remaining portions of theelectrical schematic diagrams to indicate a presence of a ring voltageon the telephone line.

The present system is also capable of sensing the caller ID informationwhich is transmitted on the telephone line between rings. Between therings, optically isolated relays U30, U31 on FIG. 7C and opticallyisolated relay U33 on FIG. 7B all operate in the period between therings so that the FSK modulated caller ID information is connected tothe CODEC and data pump DSP in FIGS. 8A and 8B, as described more fullybelow.

Referring now to FIG. 7B, more of the telephone line filtering circuitryis shown. Some of the telephone line buffering circuitry such asinductor L1 and resistor R1 are optional and are connected for varioustelephone line standards used around the word to meet localrequirements. For example, Switzerland requires a 22 millihenry inductorand 1K resistor in series the line. For all other countries, the 1Kresistor is replaced with a 0 ohm resistor.

Relay U29 shown in FIG. 7B is used to accomplish pulse dialing byopening and shorting the tip and ring wires. Optical relay X2 is engagedduring pulse dialing so that the tip and ring are shorted directly.Transistors Q2 and Q3 along with the associated discrete resistorscomprise a holding circuit to provide a current path or current loop onthe telephone line to grab the line.

FIG. 7A shows the telephone interface connections between the hardwarecomponents of the present system and the handset, headset andmicrophone.

The connections T1 and T2 for the telephone line from FIG. 7B areconnected to transformer TR1 shown in the electrical schematic diagramof FIG. 8B. Only the AC components of the signal pass throughtransformer TR1. The connection of signals attached to the secondary ofTR1 is shown for both transmitting and receiving information over thetelephone line.

Incoming signals are buffered by operational amplifiers U27A and U27B.The first stage of buffering using operational amplifier U27B is usedfor echo suppression so that the transmitted information being placed onthe telephone line is not fed back into the receive portion of thepresent system. The second stage of the input buffering throughoperational amplifier U27A is configured for a moderate amount of gainbefore driving the signal into CODEC U35.

CODEC chip U35 on FIG. 8B, interface chip U34 on FIG. 8A and digitalsignal processor (DSP) chip U37 on FIG. 8A comprise a data pump chip setmanufactured and sold by AT&T Microelectronics. A detailed descriptionof the operation of these three chips in direct connection andcooperation with one another is described in the publication entitled"AT&T V.32bis/V.32/FAX High-Speed Data Pump Chip Set Data Book"published by AT&T Microelectronics, December 1991, which is herebyincorporated by reference. This AT&T data pump chip set comprises thecore of an integrated, two-wire full duplex modem which is capable ofoperation over standard telephone lines or leased lines. The data pumpchip set conforms to the telecommunications specifications in CCITTrecommendations V.32bis, V.32, V.22bis, V.22, V.23, V.21 and iscompatible with the Bell 212A and 103 modems. Speeds of 14,400, 9600,4800, 2400, 1200, 600 and 300 bits per second are supported. This datapump chip set consists of a ROM-coded DSP16A digital signal processorU37, and interface chip U34 and an AT&T T7525 linear CODEC U35. The AT&TV.32 data pump chip set is available from AT&T Microelectronics.

The chip set U34, U35 and U37 on FIGS. 8A and 8B perform all A/D, D/A,modulation, demodulation and echo cancellation of all signals placed onor taken from the telephone line. The CODEC U35 performs DTMF tonegeneration and detection, signal analysis of call progress tones, etc.The transmission of information on the telephone line from CODEC U35 isthrough buffer U28A, through CMOS switch U36 and through line bufferU25. The CMOS switch U36 is used to switch between the data pump chipset CODEC of circuit 310 (shown in FIG. 3) and the voice control CODECof circuit 306 (also shown in FIG. 3). The signal lines AOUTN and AOUTPcorrespond to signals received from the voice control CODEC of circuit306. CODEC U35 is part of circuit 311 of FIG. 3.

The main controller of controller circuit 313 and the support circuits312, 314, 316, 317 and 308 are shown in FIGS. 5A-5C. In the preferredembodiment of the present system, the main controller is a Z80180eight-bit microprocessor chip. In the preferred implementation,microcontroller chip U17 is a Z80180 microprocessor, part number Z84CO1by Zilog, Inc. of Campbell, Calif. (also available from HitachiSemiconductor as part number HD64180Z). The Zilog Z80180 eight-bitmicroprocessor operates at 12 MHz internal clock speed by means of anexternal crystal XTAL, which in the preferred embodiment, is a 24.576MHz crystal. The crystal circuit includes capacitors C4 and C5 which are20 pf capacitors and resistor R28 which is a 33 ohm resistor. Thecrystal and support circuitry is connected according to manufacturer'sspecifications found in the Zilog Intelligent Peripheral ControllersData Book published by Zilog, Inc. The product description for theZ84CO1 Z80180 CPU from the Z84C01 Z80 CPU Product Specification pgs.43-73 of the Zilog 1991 Intelligent Peripheral Controllers databook ishereby incorporated by reference.

The Z80180 microprocessor in microcontroller chip U17 is intimatelyconnected to a serial/parallel I/O counter timer chip U15 which is, inthe preferred embodiment, a Zilog 84C90 CMOS Z80 KIOserial/parallel/counter/timer integrated circuit available from Zilog,Inc. This multi-function I/O chip U15 combines the functions of aparallel input/output port, a serial input/output port, bus controlcircuitry, and a clock timer circuit in one chip. The Zilog Z84C90product specification describes the detailed internal operations of thiscircuit in the Zilog Intelligent Peripheral Controllers 1991 Handbookavailable from Zilog, Inc. Z84C90 CMOS Z80KIO Product specification pgs.205-224 of the Zilog 1991 Intelligent Peripheral Controllers databook ishereby incorporated by reference.

Data and address buses A and B shown in FIG. 5A connect the Z80180microprocessor in microcontroller U17 with the Z80 KIO circuit U15 and agate array circuit U19, and to other portions of the electricalschematic diagrams. The gate array U19 includes miscellaneous latch andbuffer circuits for the present system which normally would be found indiscrete SSI or MSI integrated circuits. By combining a wide variety ofmiscellaneous support circuits into a single gate array, a much reduceddesign complexity and manufacturing cost is achieved. A detaileddescription of the internal operations of gate array U19 is describedmore fully below in conjunction with schematic diagrams of FIGS.10A-10C.

The memory chips which operate in conjunction with the Z80microprocessor in microcontroller chip U17 are shown in FIG. 5C. Theconnections A, B correspond to the connections to the address and databuses, respectively, found on FIG. 5A. Memory chips U16 and U13 areread-only memory (ROM) chips which are electrically alterable in place.These programmable ROMs, typically referred to as flash PROMs orProgrammable Erasable Read Only Memories (PEROMs) hold the program codeand operating parameters for the present system in a non-volatilememory. Upon power-up, the programs and operating parameters aretransferred to the voice control DSP RAM U12, shown in FIG. 9B.

In the preferred embodiment, RAM chip U14 is a pseudostatic RAM which isessentially a dynamic RAM with a built-in refresh. Those skilled in theart will readily recognize that a wide variety memory chips may be usedand substituted for pseudo-static RAM U14 and flash PROMs U16 and U13.

Referring once again to FIG. 3, the main controller circuit 313communicates with the voice control DSP of circuit 306 through dual portRAM circuit 308. The digital telephone CODEC circuit 305, the voicecontrol DSP and CODEC circuit 306, the DSP RAM 307 and the dual port RAM308 are all shown in detailed electrical schematic diagrams of FIGS.9A-9C.

Referring to FIG. 9A, the DSP RAM chips U6 and U7 are shown withassociated support chips. Support chips U1 and U2 are in the preferredembodiment part 74HCT244 which are TTL-level latches used to capturedata from the data bus and hold it for the DSP RAM chips U6 and U7.Circuits U3 and U4 are also latch circuits for also latching addressinformation to control DSP RAM chips U6 and U7. Once again, the addressbus A and data bus B shown in FIG. 9A are multi-wire connections which,for the clarity of the drawing, are shown as a thick bus wirerepresenting a grouping of individual wires.

Also in FIG. 9A, the DSP RAMs U6 and U7 are connected to the voicecontrol DSP and CODEC chip U8 as shown split between FIGS. 9A and 9B.DSP/CODEC chip U8 is, in the preferred embodiment, part number WE®DSP16C, digital signal processor and CODEC chip manufactured and sold byAT&T Microelectronics. This is a 16-bit programmable DSP with a voiceband sigma-delta CODEC on one chip. Although the CODEC portion of thischip is capable of analog-to-digital and digital-to-analog signalacquisition and conversion system, the actual D/A and A/D functions forthe telephone interface occur in digital telephone CODEC chip U12(corresponding to digital telephone CODEC circuit 305 of FIG. 3). ChipU8 includes circuitry for sampling, data conversion, anti-aliasingfiltering and anti-imaging filtering. The programmable control ofDSP/CODEC chip U8 allows it to receive digitized voice from thetelephone interface (through digital telephone CODEC chip U12) and storeit in a digitized form in the dual port RAM Chip U11. The digitizedvoice can then be passed to the main controller circuit 313 where thedigitized voice may be transmitted to the personal computer over theRS232 circuit 315. In a similar fashion, digitized voice stored by themain controller circuit 313 in the dual port RAM U11 may be transferredthrough voice control DSP chip U8, converted to analog signals bytelephone CODEC U12 and passed to the user. Digital telephone CODEC chipU12 includes a direct telephone handset interface on the chip.

The connections to DSP/CODEC chip U8 are shown split across FIGS. 9A and9B. Address/data decode chips U9 and U10 on FIG. 9A serve to decodeaddress and data information from the combined address/data bus for thedual port RAM chip U11 of FIG. 9B. The interconnection of the DSP/CODECchip U8 shown on FIGS. 9A and 9B is described more fully in the WE®DSP16C Digital Signal Processor/CODEC Data Sheet published May, 1991 byAT&T Microelectronics, which is hereby incorporated by reference.

The Digital Telephone CODEC chip U12 is also shown in FIG. 9B which, inthe preferred embodiment, is part number T7540 Digital Telephone CODECmanufactured and sold by AT&T Microelectronics. A more detaileddescription of this telephone CODEC chip U12 is described in the T7540Digital Telephone CODEC Data Sheet and Addendum published July, 1991 byAT&T Microelectronics, which is hereby incorporated by reference.

Support circuits shown on FIG. 9C are used to facilitate communicationbetween CODEC chip U12, DSP/CODEC chip U8 and dual port RAM U11. Forexample, an 8 kHz clock is used to synchronize the operation of CODECU12 and DSP/CODEC U8.

The operation of the dual port RAM U11 is controlled both by DSP U8 andmain controller chip U17. The dual port operation allows writing intoone address while reading from another address in the same chip. Bothprocessors can access the exact same memory locations with the use of acontention protocol such that when one is reading the other cannot bewriting. In the preferred embodiment, dual port RAM chip U11 is partnumber CYZC131 available from Cyprus Semiconductor. This chip includesbuilt in contention control so that if two processors try to access thesame memory location at the same time, the first one making the requestgets control of the address location and the other processor must wait.In the preferred embodiment, a circular buffer is arranged in dual portRAM chip U11 comprising 24 bytes. By using a circular bufferconfiguration with pointers into the buffer area, both processors willnot have a contention problem.

The DSP RAM chips U6 and U7 are connected to the DSP chip U8 and alsoconnected through the data and address buses to the Zilogmicrocontroller U17. In this configuration, the main controller candownload the control programs for DSP U8 into DSP RAMs U6 and U7. Inthis fashion, DSP control can be changed by the main controller or theoperating programs on the personal computer, described more fully below.The control programs stored in DSP chips U6 and U7 originate in theflash PEROM chips U16 and U17. The power-up control routine operating oncontroller chip U17 downloads the DSP control routines into DSP RAMchips U6 and U7.

The interface between the main controller circuit 313 and the personalcomputer is through SIO circuit 314 and RS232 serial interface 315.These interfaces are described more fully in conjunction with thedetailed electrical schematic diagrams of FIG. 6A-6C. RS232 connectionJ1 is shown on FIG. 6A with the associated control circuit and interfacecircuitry used to generate and receive the appropriate RS232 standardsignals for a serial communications interface with a personal computer.FIG. 6B is a detailed electrical schematic diagram showing thegeneration of various voltages for powering the hardware components ofthe electrical schematic diagrams of hardware components 20. The powerfor the present hardware components is received on connector J5 andcontrolled by power switch S34. From this circuitry of FIG. 6B, plus andminus 12 volts, plus five volts and minus five volts are derived foroperating the various RAM chips, controller chips and support circuitryof the present system. FIG. 6C shows the interconnection of the statusLED's found on the front display of the box 20.

Finally, the "glue logic" used to support various functions in thehardware components 20 are described in conjunction with the detailedelectrical schematic diagrams of FIGS. 10A-10C. The connections betweenFIGS. 10A and 10C and the previous schematic diagrams is made via thelabels for each of the lines. For example, the LED status lights arecontrolled and held active by direct addressing and data control oflatches GA1 and GA2. For a more detailed description of the connectionof the glue logic of FIGS. 10A-10C, the gate array U19 is shownconnected in FIGS. 5A and 5B.

Packet Protocol Between the PC and the Hardware Component

A special packet protocol is used for communication between the hardwarecomponents 20 and the personal computer (PC) 10. The protocol is usedfor transferring different types of information between the two devicessuch as the transfer of DATA, VOICE, and QUALIFIED information. Theprotocol also uses the BREAK as defined in CCITT X.28 as a means tomaintain protocol synchronization. A description of this BREAK sequenceis also described in the Statutory Invention Registration entitled"ESCAPE METHODS FOR MODEM COMMUNICATIONS", to Timothy D. Gunn filed Jan.8, 1993, which is hereby incorporated by reference.

The protocol has two modes of operation. One mode is packet mode and theother is stream mode. The protocol allows mixing of different types ofinformation into the data stream without having to physically switchmodes of operation. The hardware component 20 will identify the packetreceived from the computer 10 and perform the appropriate actionaccording to the specifications of the protocol. If it is a data packet,then the controller 313 of hardware component 20 would send it to thedata pump circuit 311. If the packet is a voice packet, then thecontroller 313 of hardware component 20 would distribute thatinformation to the Voice DSP 306. This packet transfer mechanism alsoworks in the reverse, where the controller 313 of hardware component 20would give different information to the computer 10 without having toswitch into different modes. The packet protocol also allows commands tobe sent to either the main controller 313 directly or to the Voice DSP306 for controlling different options without having to enter a commandstate.

Packet mode is made up of 8 bit asynchronous data and is identified by abeginning synchronization character (01 hex) followed by an ID/LIcharacter and then followed by the information to be sent. In additionto the ID/LI character codes defined below, those skilled in the artwill readily recognize that other ID/LI character codes could be definedto allow for additional types of packets such as video data, oralternate voice compression algorithm packets such as Codebook ExcitedLinear Predictive Coding (CELP) algorithm, GSM, RPE, VSELP, etc.

Stream mode is used when large amounts of one type of packet (VOICE,DATA, or QUALIFIED) is being sent. The transmitter tells the receiver toenter stream mode by a unique command. Thereafter, the transmitter tellsthe receiver to terminate stream mode by using the BREAK commandfollowed by an "AT" type command. The command used to terminate thestream mode can be a command to enter another type of stream mode or itcan be a command to enter back into packet mode.

Currently there are 3 types of packets used: DATA, VOICE, and QUALIFIED.Table 1 shows the common packet parameters used for all three packettypes. Table 2 shows the three basic types of packets with the sub-typeslisted.

                  TABLE 1                                                         ______________________________________                                        Packet Parameters                                                             ______________________________________                                        1. Asynchronous transfer                                                      2. 8 bits, no parity                                                          3. Maximum packet length of 128 bytes                                                 IDentifier byte = 1                                                           InFormation = 127                                                     4. SPEED                                                                              variable from 9600 to 57600                                                   default to 19200                                                      ______________________________________                                    

                  TABLE 2                                                         ______________________________________                                        Packet Types                                                                  ______________________________________                                        1. Data                                                                       2. Voice                                                                      3. Qualified:                                                                          a. COMMAND                                                                    b. RESPONSE                                                                   c. STATUS                                                                     d. FLOW CONTROL                                                               e. BREAK                                                                      f. ACK                                                                        g. NAK                                                                        h. STREAM                                                            ______________________________________                                    

A Data Packet is shown in Table 1 and is used for normal data transferbetween the controller 313 of hardware component 20 and the computer 10for such things as text, file transfers, binary data and any other typeof information presently being sent through modems. All packet transfersbegin with a synch character 01 hex (synchronization byte). The DataPacket begins with an ID byte which specifies the packet type and packetlength. Table 3 describes the Data Packet byte structure and Table 4describes the bit structure of the ID byte of the Data Packet. Table 5is an example of a Data Packet with a byte length of 6. The value of theLI field is the actual length of the data field to follow, not countingthe ID byte.

                  TABLE 3                                                         ______________________________________                                        Data Packet Byte Structure                                                    ______________________________________                                        byte 1       = 01h (sync byte)                                                byte 2       = ID/LI (ID byte/length indicator)                               bytes 3-127  = data (depending on LI)                                         01     ID                                                                     SYNC   LI     data    data  data data        data                             ______________________________________                                    

                  TABLE 4                                                         ______________________________________                                        ID Byte of Data Packet                                                        ______________________________________                                        Bit 7 identifies the type of packet                                           Bits 6 - 0 contain the LI or length indicator                                 portion of the ID byte                                                        7     6        5     4       3   2       1   0                                0     LI (Length Indicator) = 1 to 127                                        ______________________________________                                    

                  TABLE 5                                                         ______________________________________                                        Data Packet Example                                                           ______________________________________                                        LI (length indicator) = 6                                                     01     06                                                                     SYNC   ID    data    data  data  data  data  data                             ______________________________________                                    

The Voice Packet is used to transfer compressed VOICE messages betweenthe controller 313 of hardware component 20 and the computer 10. TheVoice Packet is similar to the Data Packet except for its length whichis, in the preferred embodiment, currently fixed at 23 bytes of data.Once again, all packets begin with a synchronization character chosen inthe preferred embodiment to be 01 hex (01H). The ID byte of the VoicePacket is completely a zero byte: all bits are set to zero. Table 6shows the ID byte of the Voice Packet and Table 7 shows the Voice Packetbyte structure.

                  TABLE 6                                                         ______________________________________                                        ID Byte of Voice Packet                                                       ______________________________________                                        7     6        5     4       3   2       1   0                                0     LI (Length Indicator) = 0                                               ______________________________________                                    

                  TABLE 7                                                         ______________________________________                                        Voice Packet Byte Structure                                                   ______________________________________                                        LI (length indicator) = 0                                                     23 bytes of data                                                              01     00                                                                     SYNC   ID     data    data  data data        data                             ______________________________________                                    

The Qualified Packet is used to transfer commands and othernon-data/voice related information between the controller 313 ofhardware component 20 and the computer 10. The various species or typesof the Qualified Packets are described below and are listed above inTable 2. Once again, all packets start with a synchronization characterchosen in the preferred embodiment to be 01 hex (01H). A QualifiedPacket starts with two bytes where the first byte is the ID byte and thesecond byte is the QUALIFIER type identifier. Table 8 shows the ID bytefor the Qualified Packet, Table 9 shows the byte structure of theQualified Packet and Tables 10-12 list the Qualifier Type byte bit mapsfor the three types of Qualified Packets.

                  TABLE 8                                                         ______________________________________                                        ID Byte of Qualified Packet                                                   ______________________________________                                        7     6        5     4       3   2       1   0                                1     LI (Length Indicator) = 1 to 127                                        ______________________________________                                    

The Length Identifier of the ID byte equals the amount of data whichfollows including the QUALIFIER byte (QUAL byte+DATA). If LI=1, then theQualifier Packet contains the Q byte only.

                  TABLE 9                                                         ______________________________________                                        Qualifier Packet Byte Structure                                               ______________________________________                                        01     85    QUAL                                                             SYNC   ID    BYTE     data data  data        data                             ______________________________________                                    

The bit maps of the Qualifier Byte (QUAL BYTE) of the Qualified Packetare shown in Tables 10-12. The bit map follows the pattern whereby ifthe QUAL byte=0, then the command is a break. Also, bit 1 of the QUALbyte designates ack/nak, bit 2 designates flow control and bit 6designates stream mode command. Table 10 describes the Qualifier Byte ofQualified Packet, Group 1 which are immediate commands. Table 11describes the Qualifier Byte of Qualified Packet, Group 2 which arestream mode commands in that the command is to stay in the designatedmode until a BREAK+INIT command string is sent. Table 12 describes theQualifier Byte of Qualified Packet, Group 3 which are information orstatus commands.

                                      TABLE 10                                    __________________________________________________________________________    Qualifier Byte of Qualified Packet: Group 1                                   7 6 5 4 3 2 1 0                                                               x x x x x x x x                                                               __________________________________________________________________________    0 0 0 0 0 0 0 0 = break                                                       0 0 0 0 0 0 1 0 = ACK                                                         0 0 0 0 0 0 1 1 = NAK                                                         0 0 0 0 0 1 0 0 = xoff or stop sending data                                   0 0 0 0 0 1 0 1 = xon or resume sending data                                  0 0 0 0 1 0 0 0 = cancel fax                                                  __________________________________________________________________________

                  TABLE 11                                                        ______________________________________                                        Qualifier Byte of Qualified Packet: Group 2                                   7    6     5     4   3   2   1   0                                            x    x     x     x   x   x   x   x                                            ______________________________________                                        0    1     0     0   0   0   0   1   = stream command mode                    0    1     0     0   0   0   1   0   = stream data                            0    1     0     0   0   0   1   1   = stream voice                           0    1     0     0   0   1   0   0   = stream video                           0    1     0     0   0   1   0   1   = stream A                               0    1     0     0   0   1   1   0   = stream B                               0    1     0     0   0   1   1   1   = stream C                               ______________________________________                                    

The Qualifier Packet indicating stream mode and BREAK attention is usedwhen a large of amount of information is sent (voice, data . . . ) toallow the highest throughput possible. This command is mainly intendedfor use in DATA mode but can be used in any one of the possible modes.To change from one mode to another, a break-init sequence would begiven. A break "AT . . . <cr>" type command would cause a change instate and set the serial rate from the "AT" command.

                  TABLE 12                                                        ______________________________________                                        Qualifier Byte of Qualified Packet: Group 3                                   7    6      5      4    3    2    1    0                                      x    x      x      x    x    x    x    x                                      ______________________________________                                        1    0      0      0    0    0    0    0   = commands                         1    0      0      0    0    0    0    1   = responses                        1    0      0      0    0    0    1    0   = status                           ______________________________________                                    

Cellular Supervisory Packet

In order to determine the status of the cellular link, a supervisorypacket shown in Table 13 is used. Both sides of the cellular link willsend the cellular supervisory packet every 3 seconds. Upon receiving thecellular supervisory packet, the receiving side will acknowledge itusing the ACK field of the cellular supervisory packet. If the senderdoes not receive an acknowledgement within one second, it will repeatsending the cellular supervisory packet up to 12 times. After 12attempts of sending the cellular supervisory packet without anacknowledgement, the sender will disconnect the line. Upon receiving anacknowledgement, the sender will restart its 3 second timer. Thoseskilled in the art will readily recognize that the timer values and waittimes selected here may be varied without departing from the spirit orscope of the present invention.

                  TABLE 13                                                        ______________________________________                                        Cellular Supervisory Packet Structure                                         ______________________________________                                        8F  ID      LI    ACK     data data        data                               ______________________________________                                    

Speech Compression

The Speech Compression algorithm described above for use in the voicemail function, the multimedia mail function and the show and tellfunction of the present system is all accomplished via the voice controlcircuit 306. Referring once again to FIG. 3, the user is talking eitherthrough the handset, the headset or the microphone/speaker telephoneinterface. The analog voice signals are received and digitized by thetelephone CODEC circuit 305. The digitized voice information is passedfrom the digital telephone CODEC circuit 305 to the voice controlcircuits 306. The digital signal processor (DSP) of the voice controlcircuit 306 is programmed to do the voice compression algorithm. Thesource code programmed into the voice control DSP is attached in themicrofiche appendix. The DSP of the voice control circuit 306 compressesthe speech and places the compressed digital representations of thespeech into special packets described more fully below. As a result ofthe voice compression algorithm, the compressed voice information ispassed to the dual port ram circuit 308 for either forwarding andstorage on the disk of the personal computer via the RS232 serialinterface or for multiplexing with conventional modem data to betransmitted over the telephone line via the telephone line interfacecircuit 309 in the voice-over-data mode of operation Show and Tellfunction 123).

Speech Compression Algorithm

To multiplex high-fidelity speech with digital data and transmit bothover the over the telephone line, a high available bandwidth wouldnormally be required. In the present invention, the analog voiceinformation is digitized into 8-bit PCM data at an 8 kHz sampling rateproducing a serial bit stream of 64,000 bps serial data rate. This ratecannot be transmitted over the telephone line. With the SpeechCompression algorithm described below, the 64 kbs digital voice data iscompressed into a 9200 bps encoding bit stream using a fixed-point(non-floating point) DSP such that the compressed speech can betransmitted over the telephone line using a 9600 baud modemtransmission. This is an approximately 7 to one compression ratio. Thisis accomplished in an efficient manner such that enough machine cyclesremain during real time speech compression to allow real time acousticand line echo cancellation in the same fixed-point DSP.

Even at 9200 bps serial data rate for voice data transmission, this bitrate leaves little room for concurrent conventional data transmission. Asilence detection function is used to detect quiet intervals in thespeech signal and substitute conventional data packets in lieu of voicedata packets to effectively time multiplex the voice and datatransmission. The allocation of time for conventional data transmissionis constantly changing depending upon how much silence is on the voicechannel.

The voice compression algorithm of the present system relies on a modelof human speech which shows that human speech contains redundancyinherent in the voice patterns. Only the incremental innovations(changes) need to be transmitted. The algorithm operates on 160digitized speech samples (20 milliseconds), divides the speech samplesinto time segments of 5 milliseconds each, and uses predictive coding oneach segment. With this algorithm, the current segment is predicted asbest as possible based on the past recreated segments and a differencesignal is determined. The difference value is compared to the storeddifference values in a lookup table or code book, and the address of theclosest value is sent to the remote site along with the predicted gainand pitch values for each segment. In this fashion, four 5 ms speechsegments can be reduced to a packet of 23 bytes or 184 bits (46 bits persample segment). By transmitting 184 bits every 20 milliseconds, aneffective serial data transmission rate of 9200 bps is accomplished.

To produce this compression, the present system includes a unique VectorQuantization (vQ) speech compression algorithm designed to providemaximum fidelity with minimum compute power and bandwidth. The VQalgorithm has two major components. The first section reduces thedynamic range of the input speech signal by removing short term and longterm redundancies. This reduction is done in the waveform domain, withthe synthesized part used as the reference for determining theincremental "new" content. The second section maps the residual signalinto a code book optimized for preserving the general spectral shape ofthe speech signal.

FIG. 11 is a high level signal flow block diagram of the speechcompression algorithm used in the present system to compress thedigitized voice for transmission over the telephone line in the voiceover data mode of operation or for storage and use on the personalcomputer. The transmitter and receiver components are implemented usingthe programmable voice control DSP/CODEC circuit 306 shown in FIG. 3.

The DC removal stage 1101 receives the digitized speech signal andremoves the D.C. bias by calculating the long-term average andsubtracting it from each sample. This ensures that the digital samplesof the speech are centered about a zero mean value. The pre-emphasisstage 1103 whitens the spectral content of the speech signal bybalancing the extra energy in the low band with the reduced energy inthe high band.

The system finds the innovation in the current speech segment bysubtracting 1109 the prediction from reconstructed past samplessynthesized from synthesis stage 1107. This process requires thesynthesis of the past speech samples locally (analysis by synthesis).The synthesis block 1107 at the transmitter performs the same functionas the synthesis block 1113 at the receiver. When the reconstructedprevious segment of speech is subtracted from the present segment(before prediction), a difference term is produced in the form of anerror signal. This residual error is used to find the best match in thecode book 1105. The code book 1105 quantizes the error signal using acode book generated from a representative set of speakers andenvironments. A minimum mean squared error match is determined in 5 mssegments. In addition, the code book is designed to provide aquantization error with spectral rolloff (higher quantization error forlow frequencies and lower quantization error for higher frequencies).Thus, the quantization noise spectrum in the reconstructed signal willalways tend to be smaller than the underlying speech signal.

The channel corresponds to the telephone line in which the compressedspeech bits are multiplexed with data bits using a packet formatdescribed below. The voice bits are sent in 100 ms packets of 5 frameseach, each frame corresponding to 20 ms of speech in 160 samples. Eachframe of 20 ms is further divided into 4 sub-blocks or segments of 5 mseach. In each sub-block of the data consists of 7 bits for the long termpredictor, 3 bits for the long term predictor gain, 4 bits for thesub-block gain, and 32 bits for each code book entry for a total 46 bitseach 5 ms. The 32 bits for code book entries consists of four 8-bittable entries in a 256 long code book of 1.25 ms duration. In the codebook block, each 1.25 ms of speech is looked up in a 256 word code bookfor the best match. The 8-bit table entry is transmitted rather than theactual samples. The code book entries are pre-computed fromrepresentative speech segments. (See the DSP Source Code in themicrofiche appendix.)

On the receiving end 1200, the synthesis block 1113 at the receiverperforms the same function as the synthesis block 1107 at thetransmitter. The synthesis block 1113 reconstructs the original signalfrom the voice data packets by using the gain and pitch values and codebook address corresponding to the error signal most closely matched inthe code book. The code book at the receiver is similar to the code book1105 in the transmitter. Thus the synthesis block recreates the originalpre-emphasized signal. The de-emphasis stage 1115 inverts thepre-emphasis operation by restoring the balance of original speechsignal.

The complete speech compression algorithm is summarized as follows:

a) Remove any D.C. bias in the speech signal.

b) Pre-emphasize the signal.

c) Find the innovation in the current speech segment by subtracting theprediction from reconstructed past samples. This step requires thesynthesis of the past speech samples locally (analysis by synthesis)such that the residual error is fed back into the system.

d) Quantize the error signal using a code book generated from arepresentative set of speakers and environments. A minimum mean squarederror match is determined in 5 ms segments. In addition, the code bookis designed to provide a quantization error with spectral rolloff(higher quantization error for low frequencies and lower quantizationerror for higher frequencies). Thus, the quantization noise spectrum inthe reconstructed signal will always tend to be smaller than theunderlying speech signal.

e) At the transmitter and the receiver, reconstruct the speech from thequantized error signal fed into the inverse of the function in step cabove. Use this signal for analysis by synthesis and for the output tothe reconstruction stage below.

f) Use a de-emphasis filter to reconstruct the output.

The major advantages of this approach over other low-bit-rate algorithmsare that there is no need for any complicated calculation of reflectioncoefficients (no matrix inverse or lattice filter computations). Also,the quantization noise in the output speech is hidden under the speechsignal and there are no pitch tracking artifacts: the speech sounds"natural", with only minor increases of background hiss at lowerbit-rates. The computational load is reduced significantly compared to aVSELP algorithm and variations of the same algorithm provide bit ratesof 8, 9.2 and 16 Kbit/s. The total delay through the analysis section isless than 20 milliseconds in the preferred embodiment. The presentalgorithm is accomplished completely in the waveform domain and there isno spectral information being computed and there is no filtercomputations needed.

Detailed Description of the Speech Compression Algorithm

The speech compression algorithm is described in greater detail withreference to FIGS. 11 through 13, and with reference to the blockdiagram of the hardware components of the present system shown at FIG.3. Also, reference is made to the detailed schematic diagrams in FIGS.9A-9C. The voice compression algorithm operates within the programmedcontrol of the voice control DSP circuit 306. In operation, the speechor analog voice signal is received through the telephone interface 301,302 or 303 and is digitized by the digital telephone CODEC circuit 305.The CODEC for circuit 305 is a companding μ-law CODEC. The analog voicesignal from the telephone interface is band-limited to about 3,500 Hzand sampled at 8 kHz by digital telephone CODEC 305. Each sample isencoded into 8-bit PCM data producing a serial 64 kb/s signal. Thedigitized samples are passed to the voice control DSP/CODEC of circuit306. There, the 8-bit μ-law PCM data is converted to 13-bit linear PCMdata. The 13-bit representation is necessary to accurately represent thelinear version of the logarithmic 8-bit μ-law PCM data. With linear PCMdata, simpler mathematics may be performed on the PCM data.

The voice control DSP/CODEC of circuit 306 correspond to the singleintegrated circuit U8 shown in FIGS. 9A and 9B as a WE® DSP16C DigitalSignal Processor/CODEC from AT&T Microelectronics which is a combineddigital signal processor and a linear CODEC in a single chip asdescribed above. The digital telephone CODEC of circuit 305 correspondsto integrated circuit U12 shown in FIG. 9(b) as a T7540 companding μ-lawCODEC.

The sampled and digitized PCM voice signals from the telephone μ-lawCODEC U12 shown in FIG. 9B are passed to the voice control DSP/CODEC U8via direct data lines clocked and synchronized to an 8 KHz clockingfrequency. The digital samples are loaded into the voice controlDSP/CODEC U8 one at a time through the serial input and stored into aninternal queue held in RAM and converted to linear PCM data. As thesamples are loaded into the end of the queue in the RAM of the voicecontrol DSP U8, the samples at the head of the queue are operated uponby the voice compression algorithm. The voice compression algorithm thenproduces a greatly compressed representation of the speech signals in adigital packet form. The compressed speech signal packets are thenpassed to the dual port RAM circuit 308 shown in FIG. 3 for use by themain controller circuit 313 for either transferring in thevoice-over-data mode of operation or for transfer to the personalcomputer for storage as compressed voice for functions such as telephoneanswering machine message data, for use in the multi-media documents andthe like.

In the voice-over-data mode of operation, voice control DSP/CODECcircuit 306 of FIG. 3 will be receiving digital voice PCM data from thedigital telephone CODEC circuit 305, compressing it and transferring itto dual port RAM circuit 308 for multiplexing and transfer over thetelephone line. This is the transmit mode of operation of the voicecontrol DSP/CODEC circuit 306 corresponding to transmitter block 1100 ofFIG. 11 and corresponding to the compression algorithm of FIG. 12.

Concurrent with this transmit operation, the voice control DSP/CODECcircuit 306 is receiving compressed voice data packets from dual portRAM circuit 308, uncompressing the voice data and transferring theuncompressed and reconstructed digital PCM voice data to the digitaltelephone CODEC 305 for digital to analog conversion and eventualtransfer to the user through the telephone interface 301, 302, 304. Thisis the receive mode of operation of the voice control DSP/CODEC circuit306 corresponding to receiver block 1200 of FIG. 11 and corresponding tothe decompression algorithm of FIG. 13. Thus the voice-control DSP/CODECcircuit 306 is processing the voice data in both directions in afull-duplex fashion.

The voice control DSP/CODEC circuit 306 operates at a clock frequency ofapproximately 24.576 MHz while processing data at sampling rates ofapproximately 8 KHz in both directions. The voicecompression/decompression algorithms and packetization of the voice datais accomplished in a quick and efficient fashion to ensure that allprocessing is done in real-time without loss of voice information. Thisis accomplished in an efficient manner such that enough machine cyclesremain in the voice control DSP circuit 306 during real time speechcompression to allow real time acoustic and line echo cancellation inthe same fixed-point DSP.

In programmed operation, the availability of an eight-bit sample of PCMvoice data from the μ-law digital telephone CODEC circuit 305 causes aninterrupt in the voice control DSP/CODEC circuit 306 where the sample isloaded into internal registers for processing. Once loaded into aninternal register it is transferred to a RAM address which holds a queueof samples. The queued PCM digital voice samples are converted from8-bit μ-law data to a 13-bit linear data format using table lookup forthe conversion. Those skilled in the art will readily recognize that thedigital telephone CODEC circuit 305 could also be a linear CODEC.

Referring to FIG. 11, the digital samples are shown as speech enteringthe transmitter block 1100. The transmitter block, of course, is themode of operation of the voice-control DSP/CODEC circuit 306 operatingto receive local digitized voice information, compress it and packetizeit for transfer to the main controller circuit 313 for transmission onthe telephone line. The telephone line connected to telephone lineinterface 309 of FIG. 3 corresponds to the channel 1111 of FIG. 11.

A frame rate for the voice compression algorithm is 20 milliseconds ofspeech for each compression. This correlates to 160 samples to processper frame. When 160 samples are accumulated in the queue of the internalDSP RAM, the compression of that sample frame is begun.

The voice-control DSP/CODEC circuit 306 is programmed to first removethe DC component 1101 of the incoming speech. The DC removal is anadaptive function to establish a center base line on the voice signal bydigitally adjusting the values of the PCM data. The formula for removalof the DC bias or drift is as follows: ##EQU1##

The removal of the DC is for the 20 millisecond frame of voice whichamounts to 160 samples. The selection of a is based on empiricalobservation to provide the best result.

Referring to FIG. 12, the voice compression algorithm in a control flowdiagram is shown which will assist in the understanding of the blockdiagram of FIG. 11. The analysis and compression begin at block 1201where the 13-bit linear PCM speech samples are accumulated until 160samples representing 20 milliseconds of voice or one frame of voice isbassed to the DC removal portion of code operating within the programmedvoice control DSP/CODEC circuit 306. The DC removal portion of the codedescribed above approximates the base line of the frame of voice byusing an adaptive DC removal technique.

A silence detection algorithm 1205 is also included in the programmedcode of the DSP/CODEC 306. The silence detection function is a summationof the square of each sample of the voice signal over the frame. If thepower of the voice frame falls below a preselected threshold, this wouldindicate a silent frame. The detection of a silence frame of speech isimportant for later multiplexing of the V-data and C-data describedbelow. During silent portions of the speech, the main controller circuit313 will transfer conventional digital data (C-data) over the telephoneline in lieu of voice data (V-data). The formula for computing the poweris ##EQU2##

If the power PWR is lower than a preselected threshold, then the presentvoice frame is flagged as containing silence (See Table 15). The160-sample silent frame is still processed by the voice compressionalgorithm; however, the silent frame packets are discarded by the maincontroller circuit 313 so that digital data may be transferred in lieuof voice data.

The rest of the voice compression is operated upon in segments wherethere are four segments per frame amounting to 40 samples of data persegment. It is only the DC removal and silence detection which isaccomplished over an entire 20 millisecond frame. The pre-emphasis 1207of the voice compression algorithm shown in FIG. 12 is the next step.The formula for the pre-emphasis is

    S(n)=S(n)-τ*S(n-1)

where τ=0.55

Each segment thus amounts to five milliseconds of voice which is equalto 40 samples. Pre-emphasis then is done on each segment. The selectionof τ is based on empirical observation to provide the best result.

The pre-emphasis essentially flattens the signal by reducing the dynamicrange of the signal. By using pre-emphasis to flatten the dynamic rangeof the signal, less of a signal range is required for compression makingthe compression algorithm operate more efficiently.

The next step in the speech compression algorithm is the long-termpredictor (LTP). The long-term prediction is a method to detect theinnovation in the voice signal. Since the voice signal contains manyredundant voice segments, we can detect these redundancies and only sendinformation about the changes in the signal from one segment to thenext. This is accomplished by comparing the linear PCM data of thecurrent segment on a sample by sample basis to the reconstructed linearPCM data from the previous segments to obtain the innovation informationand an indicator of the error in the prediction.

The first step in the long term prediction is to predict the pitch ofthe voice segment and the second step is to predict the gain of thepitch. For each segment of 40 samples, a long-term correlation lag PITCHand associated LTP gain factor B_(j) (where j=0, 1, 2, 3 correspondingto each of the four segments of the frame) are determined at 1209 and1211, respectively. The computations are done as follows.

From MINIMUM PITCH (40) to MAXIMUM PITCH (120) for indices 40 through120 (the pitch values for the range of previous speech viewed), thevoice control DSP circuit 306 computes the cross correlation between thecurrent speech segment and the previous speech segment by comparing thesamples of the current speech segment against the reconstructed speechsamples of the previous speech segment using the following formula:##EQU3## where j=40, . . . 120

S= current sample of current segment

S'= past sample of reconstructed previous segment

n_(k) =0, 40, 80, 120 (the subframe index)

and where the best fit is

    Sxy=MAX {Sxy(j)}

where J=40, . . . 120.

The value of j for which the peak occurs is the PITCH. This is a 7 bitvalue for the current segment calculated at 1209. The value of j is anindicator of the delay or lag at which the cross correlation matches thebest between the past reconstructed segment and the current segment.This indicates the pitch of the voice in the current frame. The maximumcomputed value of j is used to reduce the redundancy of the new segmentcompared to the previous reconstructed segments in the present algorithmsince the value of j is a measure of how close the current segment is tothe previous reconstructed segments.

Next, the voice control DSP circuit 306 computes the LTP gain factor Bat 1211 using the following formula in which Sxy is the current segmentand Sxx is the previous reconstructed segment: ##EQU4##

The value of the LTP gain factor B is a normalized quantity between zeroand unity for this segment where B is an indicator of the correlationbetween the segments. For example, a perfect sine wave would produce a Bwhich would be close to unity since the correlation between the currentsegments and the previous reconstructed segments should be almost aperfect match so the LTP gain factor is one.

The LTP gain factor is quantized from a LTP Gain Table. This table ischaracterized in Table 14.

                  TABLE 14                                                        ______________________________________                                        LTP Gain Quantization                                                         ______________________________________                                         ##STR1##                                                                     ______________________________________                                    

The gain value of B is then selected from this table depending uponwhich zone or range B_(segment) was found as depicted in Table 14. Forexample, if B_(segment) is equal to 0.45, then B is selected to be 2.This technique quantizes the B into a 3-bit quantity.

Next, the next LTP (Long Term Predictor) filter function 1213 iscomputed. The pitch value computed above is used to perform thelong-term analysis filtering to create an error signal e(n). Thenormalized error signals will be transmitted to the other site as anindicator of the original signal on a per sample basis. The filterfunction for the current segment is as follows:

    e(n)=S(n)-B*S'(n-pitch)

where n=0, 1, . . . 39

Next, the code book search and vector quantization function 1215 isperformed. First, the voice control DSP circuit 306 computes the maximumsample value in the segment with the formula:

    GAIN=MAX{|e(n)|}

where n=0, 1, . . . 39

This gain different than the LTP gain. This gain is the maximumamplitude in the segment. This gain is quantized using the GAIN tabledescribed in the DSP Source Code attached in the microfiche appendix.Next, the voice control DSP circuit 306 normalizes the LTP filteredspeech by the quantized GAIN value by using the maximum error signal|e(n)| (absolute value for e(n)) for the current segment and dividingthis into every sample in the segment to normalize the samples acrossthe entire segment. Thus the e(n) values are all normalized to havevalues between zero and one using the following:

    e(n)=e(n)GAIN n=0 . . . 39

Each segment of 40 samples is comprised of four subsegments of 10samples each. The voice control DSP circuit 306 quantizes 10 samples ofe(n) with an index into the code book. The code book consists of 256entries (256 addresses) with each code book entry consisting of tensample values. Every entry of 10 samples in the code book is compared tothe 10 samples of each subsegment. Thus, for each subsegment, the codebook address or index is chosen based on a best match between the10-sample subsegment and the closest 10-sample code book entry. Theindex chosen has the least difference according to the followingminimization formula: ##EQU5## where x_(i) =the input vector of 10samples, and y_(i) =the code book vector of 10 samples

This comparison to find the best match between the subsegment and thecode book entries is computationally intensive. A brute force comparisonmay exceed the available machine cycles if real time processing is to beaccomplished. Thus, some shorthand processing approaches are taken toreduce the computations required to find the best fit. The above formulacan be computed in a shorthand fashion by precomputing and storing someof the values of this equation. For example, by expanding out the aboveformula, some of the unnecessary terms may be removed and some fixedterms may be precomputed: ##EQU6## where x_(i) ² is a constant so it maybe dropped from the formula, and where the value of 1/2Σ y_(i) ² may bepre-computed and stored as the eleventh value in the code book so thatthe only real-time computation involved is the following formula:##EQU7##

Thus, for a segment of 40 samples, we will transmit 4 code book indexescorresponding to 4 subsegments of 10 samples each. After the appropriateindex into the code book is chosen, the LTP filtered speech samples arereplaced with the code book samples. These samples are then multipliedby the quantized GAIN in block 1217.

Next, the inverse of the LTP filter function is computed at 1219:

    e(n)=e(n)+B*S'(n-pitch)n=0, . . . , 39

    S'(i)=S'(n)n=40, . . . 120;i=0, . . . (120-40)

    S'(i)=e(i)i=0, . . . 40

The voice is reconstructed at the receiving end of the voice-over-datalink according to the reverse of the compression algorithm as shown asthe decompression algorithm in FIG. 13. The synthesis of FIG. 13 is alsoperformed in the compression algorithm of FIG. 12 since the past segmentmust be synthesized to predict the gain and pitch of the currentsegment.

Echo Cancellation Alqorithm

The use of the speaker 304 and the microphone 303 necessitates the useof an acoustical echo cancellation algorithm to prevent feedback fromdestroying the voice signals. In addition, a line echo cancellationalgorithm is needed no matter which telephone interface 301, 302 or303/304 is used. The echo cancellation algorithm used is an adaptiveecho canceler which operates in any of the modes of operation of thepresent system whenever the telephone interface is operational. Inparticular the echo canceller is operational in a straight telephoneconnection and it is operational in the voice-over-data mode ofoperation.

In the case of a straight telephone voice connection between thetelephone interface 301, 302, 303/304 and the telephone line interface309 in communication with an analog telephone on the other end, thedigitized PCM voice data from digital telephone CODEC 305 is transferredthrough the voice control DSP/CODEC circuit 306 where it is processed inthe digital domain and converted back from a digital form to an analogform by the internal linear CODEC of voice-control DSP/CODEC circuit306. Since digital telephone CODEC circuit 305 is a μ-law CODEC and theinternal CODEC to the voice-control DSP/CODEC circuit 306 is a linearCODEC, a μ-law-to-linear conversion must be accomplished by the voicecontrol DSP/CODEC circuit 306.

In addition, the sampling rate of digital telephone CODEC 305 isslightly less than the sampling rate of the linear CODEC of voicecontrol DSP/CODEC circuit 306 so a slight sampling conversion must alsobe accomplished. The sampling rate of digital telephone μ-law CODEC 305is 8000 samples per second and the sampling rate of the linear CODEC ofvoice control DSP/CODEC circuit 306 is 8192 samples per second.

Referring to FIG. 14 in conjunction with FIG. 3, the speech or analogvoice signal is received through the telephone interface 301, 302 or 303and is digitized by the digital telephone CODEC circuit 305 in an analogto digital conversion 1401. The CODEC for circuit 305 is a compandingμ-law CODEC. The analog voice signal from the telephone interface isband-limited to about 3,500 Hz and sampled at 8 kHz with each sampleencoded into 8-bit PCM data producing a serial 64 kb/s signal. Thedigitized samples are passed to the voice control DSP of circuit 306where they are immediately converted to 13-bit linear PCM samples.

Referring again to FIG. 14, the PCM digital voice data y(n) fromtelephone CODEC circuit 305 is passed to the voice control DSP/CODECcircuit 306 where the echo estimate signal y(n) in the form or digitaldata is subtracted from it. The substraction is done on each sample on aper sample basis.

Blocks 1405 and 1421 are gain control blocks g_(m) and g_(s),respectfully. These digital gain controls are derived from tables forwhich the gain of the signal may be set to different levels dependingupon the desired level for the voice signal. These gain levels can beset by the user through the level controls in the software as shown inFIG. 49. The gain on the digitized signal is set by multiplying aconstant to each of the linear PCM samples.

In an alternate embodiment, the gain control blocks g_(m) and g_(s) maybe controlled by Sensing the level of the speaker's voice and adjustingthe gain accordingly. This automatic gain control facilitates theoperation of the silence detection described above to assist in the timeallocation between multiplexed data and voice in the voice over datamode of operation.

In voice over data mode, the output of gain control block g_(m) isplaced in a buffer for the voice compression/decompression algorithm1425 instead of sample rate converter 1407. The samples in this mode areaccumulated, as described above, and compressed for multiplexing andtransmission by the main controller 313. Also in voice over data mode,the gain control block 1421 receives decompressed samples from the voicecompression/decompression algorithm 1425 instead of sample rateconverter 1423 for output.

The echo canceler of FIG. 14 uses a least mean square (LMS) method ofadaptive echo cancellation. The echo estimate signal subtracted from theincoming signal at 1403 is determined by function 1411. Function 1411 isa an FIR (finite impulse response) filter having in the preferredembodiment an impulse response which is approximately the length ofdelay though the acoustic path. The coefficients of the FIR filter aremodeled and tailored after the acoustic echo path of the echo takinginto account the specific physical attributes of the box that thespeaker 304 and microphone 303 are located in and the proximity of thespeaker 304 to the microphone 303. Thus, any signal placed on to thespeaker is sent through the echo cancellation function 1411 to besubtracted from the signals received by the microphone 303 after anappropriate delay to match the delay in the acoustic path. The formulafor echo replication of function box 1411 is: ##EQU8## and the result ofthe subtraction of the echo cancellation signal y(n) from the microphonesignal y(n) is

    e(n)=y(n)-y(n).

The LMS coefficient function 1413 provides adaptive echo cancellationcoefficients for the FIR filter of 1411. The signal is adjusted based onthe following formula: ##EQU9##

The echo cancellation of functions 1415 and 1417 are identical to thefunctions of 1413 and 1411, respectively. The functions 1407 and 1423 ofFIG. 14 are sample rate conversions as described above due to thedifferent sampling rates of the digital telephone CODEC circuit. 305 andthe voice control CODEC of circuit 306.

Voice Over Data Packet Protocol

As described above, the present system can transmit voice data andconventional data concurrently by using time multiplex technology. Thedigitized voice data, called V-data carries the speech information. Theconventional data is referred to as C-data. The V-data and C-datamultiplex transmission is achieved in two modes at two levels: thetransmit and receive modes and data service level and multiplex controllevel. This operation is shown diagrammatically in FIG. 15.

In transmit mode, the main controller circuit 313 of FIG. 3 operates inthe data service level 1505 to collect and buffer data from both thepersonal computer 10 (through the RS232 port interface 315) and thevoice control DSP 306. In multiplex control level 1515, the maincontroller circuit 313 multiplexes the data and transmits that data outover the phone line 1523. In the receive mode, the main controllercircuit 313 operates in the multiplex control level 1515 to de-multiplexthe V-data packets and the C-data packets and then operates in the dataservice level 1505 to deliver the appropriate data packets to thecorrect destination: the personal computer 10 for the C-data packets orthe voice control DSP circuit 306 for V-data.

Transmit Mode

In transmit mode, there are two data buffers, the V-data buffer 1511 andthe C-data buffer 1513, implemented in the main controller RAM 316 andmaintained by main controller 313. When the voice control DSP circuit306 engages voice operation, it will send a block of V-data every 20 msto the main controller circuit 313 through dual port RAM circuit 308.Each V-data block has one sign byte as a header and 23 bytes of V-data,as described in Table 15 below.

                  TABLE 15                                                        ______________________________________                                        Compressed Voice Packet Structure                                             7           6     5   4    3   2   1        0        byte                     0           0     0   0    0   0       0        0    sign                     ______________________________________                                                                P.sub.0                   β.sub.0.sup.0                                     1                                                                                  P.sub.1      β.sub.0.sup.1 2                                             P.sub.2      β.sub.0.sup.2 3                                             P.sub.3      β.sub.1.sup.0 4                                          β.sub.3    β.sub.2   β.sub.1.s                               up.2  β.sub.1.sup.1 5                                                       G.sub.1     G.sub.0    6                                                      G.sub.3     G.sub.2    7                                                        Vd.sub.0       8                                                              Vd.sub. 1       9                                                        •                                                                       •                                                                       •                                                                            Vd.sub.15       23                            ______________________________________                                        Where P.sub.n = pitch (7 bits) where n = subframe number                            β.sub.n.sup.m = Beta (3 bits)                                            G.sub.n = Gain (4 bits)                                                       Vd = Voice data (4 × 8 bits)                                      Effective Bit Rate = 184 bits / 20 msec = 9200 bps                            ______________________________________                                    

The sign byte header is transferred every frame from the voice controlDSP to the controller 313. The sign byte header contains the sign bytewhich identifies the contents of the voice packet. The sign byte isdefined as follows:

00 hex=the following V-data contains silent sound

01 hex=the following V-data contains speech information

If the main controller 313 is in transmit mode for V-data/C-datamultiplexing, the main controller circuit 313 operates at the dataservice level to perform the following tests. When the voice control DSPcircuit 306 starts to send the 23-byte V-data packet through the dualport RAM to the main controller circuit 313, the main controller willcheck the V-data buffer to see if the buffer has room for 23 bytes. Ifthere is sufficient room in the V-data buffer, the main controller willcheck the sign byte in the header preceding the V-data packet. If thesign byte is equal to one (indicating voice information in the packet),the main controller circuit 313 will put the following 23 bytes ofV-data into the V-data buffer and clear the silence counter to zero.Then the main controller 313 sets a flag to request that the V-data besent by the main controller at the multiplex control level.

If the sign byte is equal to zero (indicating silence in the V-datapacket), the main controller circuit 313 will increase the silencecounter by 1 and check if the silence counter has reached 5. When thesilence counter reaches 5, the main controller circuit 313 will not putthe following 23 bytes of V-data into the V-data buffer and will stopincreasing the silence counter. By this method, the main controllercircuit 313 operating at the service level will only provide non-silenceV-data to the multiplex control level, while discarding silence V-datapackets and preventing the V-data buffer from being overwritten.

The operation of the main controller circuit 313 in the multiplexcontrol level is to multiplex the V-data and C-data packets and transmitthem through the same channel. At this control level, both types of datapackets are transmitted by the HDLC protocol in which data istransmitted in synchronous mode and checked by CRC error checking. If aV-data packet is received at the remote end with a bad CRC, it isdiscarded since 100% accuracy of the voice channel is not ensured. Ifthe V-data packets were re-sent in the event of corruption, thereal-time quality of the voice transmission would be lost. In addition,the C-data is transmitted following a modem data communication protocolsuch as CCITT V.42.

In order to identify the V-data block to assist the main controllercircuit 313 to multiplex the packets for transmission at his level, andto assist the remote site in recognizing and de-multiplexing the datapackets, a V-data block is defined which includes a maximum of fiveV-data packets. The V-data block size and the maximum number of blocksare defined as follows:

The V-data block header=80 h;

The V-data block size=23;

The maximum V-data block size=5;

The V-data block has higher priority to be transmitted than C-data toensure the integrity of the real-time voice transmission. Therefore, themain controller circuit 313 will check the V-data buffer first todetermine whether it will transmit V-data or C-data blocks. If V-databuffer has V-data of more than 69 bytes, a transmit block counter is setto 5 and the main controller circuit 313 starts to transmit V-data fromthe V-data buffer through the data pump circuit 311 onto the telephoneline. Since the transmit block counter indicates 5 blocks of V-data willbe transmitted in a continuous stream, the transmission will stop eitherat finish the 115 bytes of V-data or if the V-data buffer is empty. IfV-data buffer has V-data with number more than 23 bytes, the transmitblock counter is set 1 and starts transmit V-data. This means that themain controller circuit will only transmit one block of V-data. If theV-data buffer has V-data with less than 23 bytes, the main controllercircuit services the transmission of C-data.

During the transmission of a C-data block, the V-data buffer conditionis checked before transmitting the first C-data byte. If the V-databuffer contains more than one V-data packet, the current transmission ofthe C-data block will be terminated in order to handle the V-data.

Receive Mode

On the receiving end of the telephone line, the main controller circuit313 operates at the multiplex control level to de-multiplex receiveddata to V-data and C-data. The type of block can be identified bychecking the first byte of the incoming data blocks. Before receiving ablock of V-data, the main controller circuit 313 will initialize areceive V-data byte counter, a backup pointer and a temporary V-databuffer pointer. The value of the receiver V-data byte counter is 23, thevalue of the receive block counter is 0 and the backup pointer is set tothe same value as the V-data receive buffer pointer. If the receivedbyte is not equal to 80 hex (80 h indicating a V-data packet), thereceive operation will follow the current modem protocol since the datablock must contain C-data. If the received byte is equal to 80 h, themain controller circuit 313 operating in receive mode will process theV-data. For a V-data block received, when a byte of V-data is received,the byte of V-data is put into the V-data receive buffer, the temporarybuffer pointer is increased by 1 and the receive V-data counter isdecreased by 1. If the V-data counter is down to zero, the value of thetemporary V-data buffer pointer is copied into the backup pointerbuffer. The value of the total V-data counter is added with 23 and thereceive V-data counter is reset to 23. The value of the receive blockcounter is increased by 1. A flag to request service of V-data is thenset. If the receive block counter has reached 5, the main controllercircuit 313 will not put the incoming V-data into the V-data receivebuffer but throw it away. If the total V-data counter has reached itsmaximum value, the receiver will not put the incoming V-data into theV-data receive buffer but throw it away.

At the end of the block which is indicated by receipt of the CRC checkbytes, the main controller circuit 313 operating in the multiplexcontrol level will not check the result of the CRC but instead willcheck the value of the receive V-data counter. If the value is zero, thecheck is finished, otherwise the value of the backup pointer is copiedback into the current V-data buffer pointer. By this method, thereceiver is insured to de-multiplex the V-data from the receivingchannel 23 bytes at a time. The main controller circuit 313 operating atthe service level in the receive mode will monitor the flag of requestservice of V-data. If the flag is set, the main controller circuit 313will get the V-data from the V-data buffer and transmit it to the voicecontrol DSP circuit 306 at a rate of 23 bytes at a time. After sending ablock of V-data, it decreases 23 from the value in the total V-datacounter.

User Interface Description

The hardware components of the present system are designed to becontrolled by an external computing device such as a personal computer.As described above, the hardware components of the present system may becontrolled through the use of special packets transferred over theserial line interface between the hardware components and the personalcomputer. Those skilled in the art will readily recognize that thehardware components of the present systems may be practiced independentof the software components of the present systems and that the preferredsoftware description described below is not to be taken in a limitingsense.

The combination of the software components and hardware componentsdescribed in the present patent application may conveniently be referredto as a Personal Communication System (PCS). The present system providesfor the following functions:

1. The control and hands-off operation of a telephone with a built-inspeaker and microphone.

2. Allowing the user to create outgoing voice mail messages with a voiceeditor, and logging incoming voice mail messages with a time and datestamp.

3. Creating queues for outgoing faxes including providing the abilityfor a user to send faxes from unaware applications through a printcommand; also allowing the user the user to receive faxes and loggingincoming faxes with a time and date stamp.

4. Allowing a user to create multi-media messages with the messagecomposer. The message can contain text, graphics, picture, and soundsegments. A queue is created for the outgoing multi-media messages, andany incoming multi-media messages are logged with a time and date stamp.

5. Providing a way for a user to have a simultaneous data and voiceconnection over a single communication line.

6. Providing terminal emulation by invoking an external terminalemulation program.

7. Providing address book data bases for all outbound calls and queuesfor the telephone, voice mail, fax manager, multi-media mail andshow-and-tell functions. A user may also search through the data baseusing a dynamic pruning algorithm keyed on order insensitive matches.

FIG. 16 shows the components of a computer system that may be used withthe PCS. The computer includes a keyboard 101 by which a user may inputdata into a system, a computer chassis 103 which holds electricalcomponents and peripherals, a screen display 105 by which information isdisplayed to the user, and a pointing device 107, typically a mouse,with the system components logically connected to each other viainternal system bus within the computer. The PCS software runs on acentral processing unit 109 within the computer.

FIG. 17 reveals the high-level structure of the PCS software. A mainmenu function 111 is used to select the following subfunctions: setup113, telephone 115, voice mail 117, fax manager 119, multi-media mail121, show & tell 123, terminal 125, and address book 127.

The preferred embodiment of the present system currently runs underMicrosoft Windows® software running on an IBM® personal computer orcompatible. However, it will be recognized that other implementations ofthe present inventions are possible on other computer systems andwindowing software without loss of scope or generality.

FIG. 18 describes the control structure of the main menu 111 in greaterdetail. A timer 131 sends a timing signal to a control block 129 inorder to make the control block 129 active substantially once every 10seconds. It will be recognized that other timing intervals may be usedas appropriate to the windowing system being used without loss ofgenerality. A status 133 is used to preclude other applications orprogram blocks from taking control of a communications port byindicating that the port is currently being used by the PCS. Thecontroller 129 looks at all outbound queues in voice mail 117, faxmanager 119, and multi-media mail 121, and if there is an outgoingmessage in one of the outbound queues, initiates a dispatch. A signal isthen sent to the status box in order to preclude other applications orprogram blocks from using the serial communications port.

The control block 129 also monitors incoming calls and invokes theappropriate program block, either voice mail 117, fax manager 119,multi-media mail 121, or show & tell 123, in order to further processthe incoming call. Additionally, the control block 129 is used to invoketelephone functions 115, terminal emulation functions 125, and allowusers to edit the data base of addresses with the address book function127. The control block 129 further provides for the initialization ofPCS parameters via the setup function 113. The main menu, as it isdisplayed to the user, is shown in FIG. 2.

FIG. 19 illustrates the structure of control block 129. Upon selectingthe setup function 113, the user has access to initialization functions135 which include serial port, answer mode, hold call, voice mail, PBX,fax, multi-media mail, and show & tell initializations. Upon selectingtelephone 115, the user has access to telephone functions 137 whichinclude equipment select, volume control, and call functions as shown inthe screen display of FIG. 49. Upon selecting voice mail 117, voice mailfunctions 139 are provided which include a voice editor, voice messagesto be sent, and voice messages received as shown in the screen displayof FIG. 50. Upon selecting fax manager 119, fax manager functions 141are provided which include setup functions, faxes to be sent, and faxesto be received as shown in the screen display of FIG. 52. If multi-mediamail 121 is selected, multi-media mail functions 143 are provided whichinclude the setup function, multi-media messages to be sent, andmulti-media messages received function as illustrated by the screendisplay shown in FIG. 53. If show & tell 123 is selected, show & tellfunctions 145 are provided to the user which include open, select new,and help as illustrated in the screen display of FIG. 54. If theterminal function 125 is selected by the user, the terminal emulationfunction 147 is provided to the user via a terminal emulation block. Ifaddress book 127 is selected, address book functions 149 are provided tothe user which include file functions, edit functions, and helpfunctions as shown in the screen display of FIG. 55.

The setup functions 135 are accessed by an initialization menu as shownin the screen display of FIG. 40. The PCS software provides support forany communications port that is contained within the personal computer.FIG. 41 shows the screen display shown to a user to enable a user toselect a specific communications port to be used by the PCS software. Auser may also specify what action is to be taken by the PCS softwarewhen an incoming call arrives. The screen display of FIG. 42 is shown tothe user while FIG. 20 describes the full control of the answer modesetup initialization procedure. Upon selecting answer mode setup 151,the user is presented with eight answer mode choices 153 and describedas follows:

1. PCS does not answer. The PCS software does not answer an incomingcall and the telephone equipment acts as normal.

2. Voice mail. The PCS software answers the incoming call and acts as ananswering machine to record messages.

3. Fax. The PCS software answers the incoming calls and acts as a faxmachine.

4. Multi-media mail. The PCS software answers the incoming call andreceives multi-media mail that is being sent by a remote caller.

5. Show & tell. The PCS software enables simultaneous data and voicecommunication using the same communication line.

6. Terminal. The PCS provides terminal emulation through a terminalemulation block, or optionally transfers control to a third partyterminal emulation program.

7. Automatic. The incoming call is analyzed and the appropriate mode isautomatically entered.

The user may additionally enter a numeric value which represents thenumber of rings to wait before the PCS software answers an incomingcall.

If from the setup functions 135 the hold call function is selected, thehold call display as illustrated in FIG. 43 is shown to the user, whomay then enter a numeric value to specify the number of hours alloutgoing calls are to be held. If at setup functions 135 the voice mailsetup option is selected, the screen display as illustrated in FIG. 44is displayed to the user who may then enter a file name to be used as agreeting file for all incoming calls. If at setup functions 135 the PBXsetup function is selected, the screen display of FIG. 45 is shown tothe user who may then enter a dialing prefix to be applied to anyoutgoing telephone number. This provides for an easy way to use a listof telephone numbers with an in-house PBX system without makingmodification to the telephone number list. If at setup functions 135 thefax setup function is selected, the screen display of FIG. 46 isdisplayed to the user who may then enter a numeric value whichrepresents the number of times to attempt to send a fax before timingout. If at setup functions 135 the multi-media mail setup function isselected, the display of FIG. 47 is shown to the user who may then enterthe name of a file to be used for user information. If at setupfunctions 135 the show & tell function is selected, the screen displayof FIG. 48 is shown to the user who may then enter show & tell userinformation.

FIG. 49 shows the telephone control function display as shown to theuser. FIG. 21 further illustrates the steps and options 155 availablewhen the telephone control function 115 is selected. The user may usethe mouse to select between a speaker phone, a handset, or a headset tobe used with the communications device, and may adjust the volume of thespeaker with a first logical slider switch, and the gain of themicrophone with a second logical slider switch, both slider switchesbeing displayed on the screen. During a call, the user may select tosave a telephone number, redial a telephone number, record a call, flashbetween calls, mute a call, or place a call on hold.

FIG. 22 shows the voice mail functions 157 that are available uponselecting voice mail 117. A user may set up a voice mail greeting fileor may edit a voice mail message by selecting the voice mail editor asshown in FIG. 50. The PCS software provides for two voice mail queues:the first for voice mail messages to be sent and the second for voicemail messages received. In the send queue a user may add messages to thequeue, listen to messages in the queue, delete messages from the queue,or refresh the send queue display. With the receive queue a user maylisten to messages in the queue, store messages in the queue, deletemessages from the queue, forward messages from the queue to anothervoice mail user, or refresh the queue. The voice mail editor as shown tothe user in FIG. 51 allows the user to select file functions to open orsave a voice mail message, edit functions for editing a voice mailmessage, playing, recording, and pausing a voice mail message, andadjusting the play volume and the record volume. The user may alsooptionally select between a speaker phone, headset, and handset.

FIG. 52 illustrates the fax manager display as shown to the user. FIG.23 illustrates the fax manager functions 159 that are available to auser after selecting the fax manager function 119 from the main menu111. The fax manager function provides for two queues: the first forfaxes to be sent and the second for faxes that are received. Whenreviewing the first queue of faxes to be sent, the user may preview afax, print a fax, delete a fax from the queue, refresh the send faxqueue, or forward a fax to another user. When reviewing the second queueof received faxes, the user may view a fax, print a fax, delete a fax,refresh the received fax queue, forward a fax to another user, or storea fax.

FIG. 53 describes the multi-media mail display that is shown to theuser. FIG. 24 describes the multimedia mail functions 161 that areavailable to a user upon selecting the multi-media mail function 121.Upon selecting multi-media mail the PCS software provides for setup,composing a message with the message composer, or allowing the user toview and edit multi-media messages in two queues: the first queue havingmulti-media messages to be sent and the second queue having multi-mediamessages that have been received. When reviewing the first queue of sendmessages, a user may add messages to the queue, preview messages in thequeue, delete messages from the queue, or change attributes of messagesin the queue. When a user is accessing the second queue of multi-mediamessages that have been received, the user may view messages, storemessages, delete messages, forward messages to another user, or refreshthe queue.

FIG. 54 illustrates the display shown to the user upon selecting theshow & tell function 123 from the main menu 111. FIG. 55 illustrates thedisplay that is shown to the user upon selecting the address bookfunction 127 from the main menu 111. A user may open a previously storedaddress book file or may edit an existing address book file by adding,deleting, or changing entries that are in the file. Additionally, thePCS software provides for a user to search through the data base byusing a dynamic pruning algorithm keyed on order insensitive matches. Asthe user enters a search string at a dialogue box, the list of matchesdisplayed is automatically updated in real time to correspond to as muchof the search string that has already been entered. The list continuesto be updated until the search string has been completely entered.

Software Control Description

The preferred embodiment of the software control system of the presentinvention runs under Microsoft Windows software on an IBM PC orcompatible. It will be recognized that other software implementationsare available on other types of computers and windowing systems withoutloss of generality.

FIG. 25 shows the timing loop 131 of FIG. 18 in greater detail. In orderto process pending actions, the timer 131 checks the fax out queue at163, the multi-media out queue at 165, the voice mail out queue at 167,and the communication port at 169. The three output queues and thecommunication port are checked substantially once every 10 seconds todetermine if there are any pending actions to be performed. If the timerdoes find a pending action in one of the queues, or if there isinformation coming in from the communication port, a secondary timer ofduration 100 miliseconds is spawned in order to handle each pendingaction. This polling of the output queues continues as long as the mainPCS software is active and running. The polling interval rate ofsubstantially 10 seconds is short enough such that there is nosignificant time delay in handling any pending action in any of theoutput queues. It will be recognized that short timing intervals otherthan substantially 10 seconds may be used without loss of generality.

FIG. 38 illustrates the flow control for the outgoing queue timer. Theincoming timer is stopped at 3701 and the communication port isinitialized at 3703. A job record 3707 is used to determine thedestination for this message and dial the telephone at 3705. After aprotocol handshake at 3709, an output page is sent at 3711 which mayinclude a fax code file 3713. The remaining number of pages to be sentis checked at 3715, and if there are more pages, control returns to 3709so that the additional page or pages can be sent to the destination.Otherwise, if at 3715 there are no more pages to be sent, the job record3707 is updated at 3717 and the incoming time is restarted at 3719.

FIG. 39 illustrates the flow control for the incoming queue timer. At3801 the communication port is initialized, and the software waits for aring indicator from an incoming call at 3803. After receiving a ring,the software answers, establishes a connection via a protocol handshakeat 3805, and receives input data at 3807. If at 3809 there is more datato be received, control passes back to 3805 so that the pending data maybe received. Otherwise, if at 3809 there is no more data to be received,the call is terminated at 3811.

FIG. 26 shows the control software used for a hands-off telephone. Thetelephone control software is invoked upon selection of the telephoneoption 115 shown in FIG. 2. Upon selection of the telephone option at2501, any timers that have been spawned or are currently running aredisabled, and at 2503 the communication port for the personal computeris initialized. At 2505 the telephone control software handles anyoptions that have been selected by the user as displayed to the user byFIG. 49. At 2507 the user may logically select between a headset,handset, or speaker phone, at 2509 the user may adjust the volume levelof the speaker or the gain of the microphone, and at 2511 the user mayselect between mute, hold, record, and redial functions for thehands-off telephone. After the user selects options at 2513, a telephonenumber is dialed and the call initiated. After the call is complete, at2515 the user hangs up, and at 2517 any timers that had been stopped at2503 are restarted and at 2519 the communication port is restored to itsprevious state.

FIG. 27 shows the voice mail control software that is invoked by option117 from FIG. 2. The user may select either a new file or record at2601, open an existing file at 2613, or abort at 2625. Upon selecting afile or record at 2603, the file may be saved at 2611 or the user mayselect options at 2605. The selectable options include setting thevolume or record levels at 2607, or selecting between a handset, aheadset, or a speaker phone at 2609 as shown to the user by the voicemail editor display given in FIG. 51. If an existing file is opened at2613, the file name is selected at 2615 whereupon a user may then playthe file at 2617 or select options at 2619. The selectable options from2619 include setting the volume or record levels at 2621 or selectingbetween a headset, handset, or speaker phone at 2623. Once a voice mailmessage has been recorded or opened from a previous session, a graphicalrepresentation of the voice mail message is displayed in a window with xand y dimensions, where the x dimension represents time and the ydimension represents the volume of the voice mail message at that pointin time. The pointing device may be used to modify the voice message bygraphically changing the two dimensional voice message plot. The cursoris placed within the two dimensional voice message plot in order toindicate the portion of the voice message to be modified. A scrollbutton beneath the two dimensional voice message plot may be used toselect the time portion of the message to be displayed within the twodimensional plot window. Time is shown increasing from the left to theright, corresponding to the x axis of the plot. As the scroll button ismoved to the right, later portions of the voice message are displayed.As the scroll button is moved to the left, earlier portions of the voicemail message are displayed. A numeric value is shown substantially onthe left side of the two dimensional plot window which is updatedcontinuously and corresponds to the time value of the current locationon the x axis.

Upon the recording of a voice mail message, the voice mail may be addedto the voice mail send queue as displayed to the user in FIG. 50 anddescribed at 159 in FIG. 23. Upon adding a voice mail message to thevoice mail queue, the user is prompted as shown in FIG. 56 to enter aname and a telephone number to whom the voice mail message must be sent.The user may select from a predetermined list of voice mail recipientspreviously set up in the address book as shown to the user in FIG. 55.

FIGS. 28 and 29 show how the fax code drivers of the PCS softwaretypically work. The fax capability is tied to the windowing system printcommand so that facsimile transmissions may be sent by any software thatcan print through the windowing environment. At 2701 in FIG. 28, a highresolution fax driver examines a print file at 2703 that has beenprinted by a windowing system application. The print file is thenconverted and imaged at 2705 into a PCX bit-mapped format which of thecorrect horizontal and vertical resolution in dots per inch (dpi) forhigh resolution facsimile devices. FIG. 29 shows an equivalent processused by a low resolution fax driver at 2801. A print file 2803 isconverted at 2805 to a low resolution PCX bit-mapped format for use withlow resolution facsimile devices. Upon converting a print file to afacsimile document, the facsimile document may be added to the fax outqueue as displayed to the user in FIG. 52. The user may select the nameand telephone number of a person to send the fax to through the addressbook function as shown to the user in FIG. 55. Received faxes may beviewed or printed from the fax in queue.

FIG. 30 shows the multi-media control software that is invoked by a userselecting multi-media option 121 shown to a user in FIG. 2. The user mayinitialize multi-media mail settings at 2901, select various multi-mediamail options at 2905, or invoke a message composer at 2903 as shown tothe user by the screen display of FIG. 57. In the message composer 2903,the user may open a new file at 2907 or cancel the message composer at2911 or open an existing file at 2909. Upon opening an existing file,the file name is selected is 2913 and the multi-media mail editor 2917is invoked at 2915. While editing a file at 2919 a user may select toalternately play a message at 2921 or record a message at 2923. The usermay edit in line mode either inserting, deleting, joining, or splittinglines, edit in block mode by moving, copying, deleting, or highlightingblocks of text, changing the font used to display the text, or changingthe indent and justification attributes of paragraphs of text. Standardsearch features such as forward and backward search and global searchand replace are also available through an "other" menu. FIG. 31 furtherdescribes the options available that are shown to the user by themulti-media edit display of FIG. 57 at 3001 the file line edit, blockedit, fonts, paragraph, voice, "other", and help options are available.If at 3001 the user selects "file", the options shown in 3003, save,save as, page layout, printer setup, and print, are available to theuser. If at 3001 the user selects "voice", the options available at3005, record voice, stop recording, play voice, stop play, store voiceto disk, and get voice from disk, are available to the user. Afterselecting from edit controls 3001, at 3007 the appropriate action istaken by the PCS software, the voice icon is displayed, and anyadditional graphics are also displayed.

After a multi-media message has been created, at 3101 the user may add amessage to the send queue. Upon adding the message, at 3103 the userselects the name and telephone number of a person to send a message to,or at 3105 selects a name from the address book and at 3107 selects adestination from the address book to send the message to. The message isthen added to the job list at 3109 at 3111 the job scroll list isupdated.

FIG. 33 shows the options available when a multi-media message has beenreceived. At 3201 the user may select and view the message. At 3203 thesoftware selects the appropriate message, opens the job list at 3205,loads the received message at 3207 and invokes the message composer at3209 in order to display the received multi-media message.

FIG. 34 shows the software control procedure used with the show and tellfeature to provide data over voice capability when selected at 123 fromFIG. 2. At 3301 any existing timers are disabled and the communicationsport is initialized at 3303. The destination for any messages isaccessed from the address book at 3305, whereupon the telephone numberis dialed and the connection is set up at 3307. Upon detecting asuccessful connection, the data over voice mode is initialized at 3309while a message is being transmitted at 3311 the user may select optionsof either quitting PCS at 3315, or invoking a terminal emulation at3313. The data over voice connection is accomplished by multiplexing thebandwidth of the connection as described elsewhere in thisspecification.

FIG. 35 shows the control procedure used when receiving data over voicemessages. At 3401 any existing timers are disabled and thecommunications port is initialized at 3403. At 3405 the software waitsfor a ring indicator. If, after a predetermined amount of time, noringing indicator is detected, a timeout occurs at 3407 whereupon thePCS software aborts are returns at 3409. Otherwise, at 3411 a ringindicator is received, the data over voice connection is established at3413, and the user may select options at 3415 either to quit the PCSsoftware at 3419 and close the communication port at 3421, or invoketerminal emulation at 3417.

FIG. 36 shows the control procedure used when transmitting voice mailmessages. At 3501 a voice mail message is added to the send queue,whereupon at 3503 the user specifies the message file name, and at 3505indicates a name from a previously entered address book entry 3505 anddestination telephone number 3507 to send the message to. The message isthen added to the Job list at 3509, and the queue display is updated at3511.

FIG. 37 shows the control procedure used when receiving voice mailmessages. At 3601 a voice mail message is received and recorded by thePCS software. At 3603 the user selects a message to be reviewed oredited, after which the software opens the Job list at 3605, loads theselected recorded message into memory at at 3607, and invokes the voiceeditor at 3609 with the selected recorded message. The user may thenselect various functions from within the voice editor to review or editthe message as previous described above.

Data Structures Description

Descriptions of the data structures and variable names and types aregiven below for the preferred embodiment of the present invention. Thepreferred embodiment is written the in C programming language runs underMicrosoft Windows software on an IBM PC or compatible system, however,it will be recognized that these data structures and methods are genericand potentially useful for a wide variety of other windowing software,systems, and programming languages.

Address Book

Address key types are used to indicate how information within theaddress book should be displayed to the user:

    ______________________________________                                        KEY.sub.- NAME  0        list addresses by name                               KEY.sub.- AFFILIATION                                                                         1        by affiliation                                       KEY.sub.- STREETADRS                                                                          2        by streetadrs code                                   KEY.sub.- CITYSTATE                                                                           3        by citystate code                                    KEY.sub.- ZIP   4        by zip code                                          KEY.sub.- PHONE 5        by phone code                                        KEY.sub.- FAX   6        by fax code                                          KEY.sub.- MISC  7        by misc code                                         KEY.sub.- TYPE.sub.- COUNT                                                                    8        number of key types                                  ______________________________________                                    

The address entry structure defines what fields are associated with eachaddress book entry:

    ______________________________________                                        s.sub.- address.sub.- entry.sub.- struct                                      unsigned int delFlag ;                                                                         delete flag                                                  unsigned int caName ;                                                                          Name field                                                   unsigned int caAffiliation ;                                                                   Affiliation field                                            unsigned int caStreetAdrs ;                                                                    Street Address field                                         unsigned int caCityState ;                                                                     City State field                                             unsigned int caZip ;                                                                           Zip field                                                    unsigned int caPhone ;                                                                         Phone Number field                                           unsigned int caFax ;                                                                           Fax Number field                                             unsigned int caMisc ;                                                                          Miscellaneous field                                          Key Names[KEY.sub.- TYPE.sub.- COUNT] =                                       "Name",                                                                       "Affiliation",                                                                "StreetAdrs",                                                                 "CityState",                                                                  "Zip",                                                                        "Phone",                                                                      "Fax",                                                                        "Misc"                                                                        ______________________________________                                    

The following character strings are used to hold address bookinformation:

    __________________________________________________________________________    char  g.sub.- adrsbooktemp[MAX.sub.- FILE.sub.- NAME.sub.- LEN[ =                   "temp.adr";                                                             static                                                                              char g.sub.- ClipboardFormat[] = "CF.sub.- FLOCOM" ;                    static                                                                              char g.sub.- adrsbookDlgName[] = "AdrsBkDlg" ;                          static                                                                              char g.sub.- adrsbookFileName[MAX.sub.- FILE.sub.- NAME.sub.- LEN]      static                                                                              char g.sub.- FaxStr[ADDRESS.sub.- ENTRY.sub.- FIELD.sub.- SIZE];        static                                                                              char g.sub.- PhoneStr[ADDRESS.sub.- ENTRY.sub.- FIELD.sub.- SIZE]             ;                                                                       static                                                                              char g.sub.- NameStr[ADDRESS.sub.- ENTRY.sub.- FIELD.sub.- SIZE]        __________________________________________________________________________          ;                                                                   

Fax Send and Receive Queues

A structure definition is used to indicate how information within thefax send and receive queues is stored:

    ______________________________________                                        struct s.sub.- jobDesc {                                                      union {                                                                       unsigned char phoneNum[32];                                                   short nbrOfJobs;                                                              } ul;                                                                         union {                                                                       short nextJobNum;                                                             unsigned char date[16];                                                       } u2;                                                                         unsigned char title[64];                                                      unsigned char receiver[32];                                                   unsigned char coverFile[32];                                                  unsigned char telNum[32];                                                     unsigned char faxCdFile[8];                                                   unsigned char time[8];                                                        WORD zState;                                                                  time.sub.- t zTime;                                                           short zPages;                                                                 short zResult;                                                                int zTries;                                                                   char zType;                                                                   char zDummy[19];                                                              } t.sub.- jobDesc, *tp.sub.- jobDesc, far *tpl.sub.- jobDesc;                 static t.sub.- jobDesc g.sub.- curJob, g.sub.- bufJob;                        ______________________________________                                    

Multi Media Send and Receive Queues

The following structure definition is used to indicate how informationwithin the multi media send and receive queues is stored:

    ______________________________________                                        struct s.sub.- jobDesc {                                                      union {                                                                       unsigned char phoneNum[32];                                                   short    nbrOfJobs;                                                           } ul;                                                                         union {                                                                       short nextJobNum;                                                             unsigned char date[16];                                                       } u2;                                                                         unsigned char title[64];                                                      unsigned char receiver[32];                                                   unsigned char coverFile[32];                                                  unsigned char telNum[32];                                                     unsigned char MMCdFile[8];                                                    unsigned char time[8];                                                        WORD zState;                                                                  time.sub.- t zTime;                                                           short zPages;                                                                 short zResult;                                                                int zTries;                                                                   char zType;                                                                   char zDummy[19];                                                              } t.sub.- jobDesc, *tp.sub.- jobDesc, far *tpl.sub.- jobDesc;                 ______________________________________                                    

Show and Tell

The show and tell structure definition is the same as that used in theaddress book. The static variables to define field sizes are givenbelow:

    ______________________________________                                        /*                                                                            Static Variables Used For Address Book Proc. The Variable                     Names Are Same As The Ones In "Adrsbook" , So That Same                       Modules Could Be Used                                                         */                                                                            #define ADDRESS.sub.- ENTRY.sub.- FIELD.sub.- SIZE                                                     64                                                   #define NUM.sub.- MEMBERS.sub.- ADRS.sub.- STRUCT                                                       8                                                   #define NUM.sub.- ADRS.sub.- FIELDS                                                                     8                                                   #define VAL.sub.- PAUSE 2000                                                  #define PCKT.sub.- COM.sub.- TIME                                                                      20                                                   #define WAIT.sub.- RING.sub.- TIME                                                                     120                                                  #define DIAL.sub.- TIME  60                                                   //1029 vasanth for delay before %p1 command                                   #define DELAYTIME        10                                                   /* Address Key Types */                                                       #define KEY.sub.- NAME                                                                         0 /* list addresses by name                                  */                                                                            #define KEY.sub.- AFFILIATION                                                                  1 /* by affiliation */                                       #define KEY.sub.- STREETADRS                                                                   2 /* by streetadrs code */                                   #define KEY.sub.- CITYSTATE                                                                    3 /* by citystate code */                                    #define KEY.sub.- ZIP                                                                          4 /* by zip code */                                          #define KEY.sub.- PHONE                                                                        5 /* by phone code */                                        #define KEY.sub.- FAX                                                                          6 /* by fax code */                                          #define KEY.sub.- MISC                                                                         7 /* by misc code */                                         #define KEY.sub.- TYPE.sub.- COUNT                                                             8 /* number of key types */                                  ______________________________________                                    

Voice Mail Send and Receive Queues

The voice mail structure definition and static variables to define fieldsizes are given below:

    __________________________________________________________________________    #define READCOUNT                                                                            24000                                                          #define FRAME.sub.- SIZE                                                                     2000                                                           #define SCROLL.sub.- STEP                                                                    4                                                              #define POS.sub.- IN.sub.- PAGE                                                              100                                                            #define T.sub.- POSITION                                                                     0.02                                                           #define COMP.sub.- FRAME.sub.- SZ                                                            24                                                             #define BYTES.sub.- IN.sub.- FRAME                                                           12000                                                          #define BYTE.sub.- NUMBER                                                                    6                                                              #define WAVE.sub.- COEF                                                                      1                                                              #define FORMAT.sub.- STR1                                                                    "%s -> %s # %s Schedule: %s: %s"                               #define FORMAT.sub.- STR2                                                                    "%s -> %s # %s Sent: %s: %s"                                   #define OUT.sub.- JOBS.sub.- FILE.sub.- NAME "jobs"                           #define IN.sub.- JOBS.sub.- FILE.sub.-NAME "jobs"                             #define MAX.sub.- JOBS                                                                   10                                                                 struct s.sub.- jobDesc {                                                      union {                                                                       unsigned char phoneNum[32];                                                   short nbrOfJobs;                                                              } ul;                                                                         union {                                                                       short nextJobNum;                                                             unsigned char date[16];                                                       } u2;                                                                         unsigned char title[64];                                                      unsigned char receiver[ 32];                                                  unsigned char coverFile[32];                                                  unsigned char telNum[32];                                                     unsigned char faxCdFile[8];                                                   unsigned char time[8];                                                        WORD zState;                                                                  time.sub.- t zTime;                                                           short zPages;                                                                 short zResult;                                                                int zTries;                                                                   char zType;                                                                   char zDummy[19];                                                              } t.sub.- jobDesc, *tp.sub.- jobDesc, far *tpl.sub.- jobDesc;                 static HWND hwndVMD1g;                                                        static int g.sub.- cxWave ; /* width of waveform window */                    static int g.sub.- cyWave ; /* height of waveform window*/                    static int g.sub.- nSamples ; /* sample counter */                            static char g.sub.- outVMDir[MAX.sub.- FILE.sub.- NAME.sub.- LEN];            static char g.sub.- inVMDir[MAX.sub.- FILE.sub.- NAME.sub.- LEN];             static HANDLE g.sub.- outJobsHnd1 = 0;                                        static HANDLE g.sub.- inJobsHnd1 = 0;                                         static HANDLE g.sub.- jobFileHnd1 = 0;                                        static t.sub.- jobDesc g.sub.- outJobs0;                                      static t.sub.- jobDesc g.sub.- outJobsn;                                      static t.sub.- jobDesc g.sub.- inJobs0;                                       static t.sub.- jobDesc g.sub.- inJobsn;                                       static short g.sub.- numOfOutJobs =0;                                         static short g.sub.- numOfInJobs =0;                                          static int g.sub.- VMOutIx = -1;                                              static int g.sub.- VMInIx = -1;                                               static OFSTRUCT jOfStruct ;                                                   static FILE *vdata ;                                                          static OFSTRUCT OfStruct ;                                                    static int Hfile ;                                                            static OFSTRUCT oOfStruct ;                                                   static int hoFile ;                                                           static char g.sub.- inFileName[MAX.sub.- FILE.sub.- NAME.sub.- LEN] ;         static int hjFile ;                                                           static char g.sub.- jFileName[MAX.sub.- FILE.sub.- NAME.sub.- LEN] ;          static char g.sub.- uFileName[MAX.sub.- FILE.sub.- NAME.sub.- LEN] ;          static char g.sub.- oFileName[MAX.sub.- FILE.sub.- NAME.sub.- LEN] ;          static char g.sub.- sendListenFile[MAX.sub.- FILE.sub.- NAME.sub.- LEN]       static char g.sub.- recListenFile[MAX.sub.- FILE.sub.- NAME.sub.- LEN]        __________________________________________________________________________

The following are variables to be used to write the playback or recordlevel into the pcs.ini volume field names:

char *g₋₋ PlayVolume="Play Volume";

char *g₋₋ RecVolume="Record Volume";

Default volume levels for record and play:

    ______________________________________                                        int g.sub.- recPos = 5;                                                       int g.sub.- playPos = 5;                                                      char g.sub.- MicroVol[10] = ">MV0";                                                              Microphone level to be                                                        used for recording.                                        char g.sub.- PlayVol[10] = ">SV0";                                                               Speaker level to be used                                                      for play.                                                  int g.sub.- offhk = FALSE;                                                                 off hook flag indicating that                                                 either record or play is in                                                   progress. If flag is set only then                                            send packet commands to                                                       increase/decrease volume level.                                  ______________________________________                                    

Below are given Static Variables Used For Address Book Proc. TheVariable Names Are Same As The Ones In "Adrsbook.c", So That SameModules Could Be Used.

    ______________________________________                                        #define MAX.sub.- REC.sub.- VOL                                                                         13                                                  #define MIN.sub.- VOL.sub.- LEVEL                                                                        0                                                  #define MAX.sub.- PLAY.sub.- VOL                                                                         9                                                  #define MAX.sub.- ADRS.sub.- ENTRIES                                                                    512                                                 #define ADDRESS.sub.- ENTRY.sub.- FIELD.sub.- SIZE                                                      64                                                  #define NUM.sub.- MEMBERS.sub.- ADRS.sub.- STRUCT                                                        8                                                  #define NUM.sub.-ADRS.sub.- FIELDS                                                                       8                                                  ______________________________________                                    

The present inventions are to be limited only in accordance with thescope of the appended claims, since others skilled in the art may deviseother embodiments still within the limits of the claims.

Microfiche Appendix

The microfiche appendix to the present patent application contains thesource code for the software running on the personal computer and thesource code for the software running on the voice control DSP/CODEC.

We claim:
 1. A communication module for use with a personal computer,comprising:communications interface means connected for communicating tothe personal computer for transferring data between the personalcomputer and the communications module; telephone line interface meansfor connection to a telephone line and for full duplex digitalcommunication over the telephone line; voice interface means forreceiving local voice signals from a local user and for conveying remotevoice signals from a remote user to the local user; full-duplexconversion means connected to the voice interface means for convertingthe local voice signals into outgoing digital voice data and forconverting incoming digital voice data into the remote voice signals;digital signal processor means connected to the full-duplex conversionmeans for compressing the outgoing digital voice data into compressedoutgoing digital voice data packets and for decompressing compressedincoming digital voice data packets into the incoming digital voicedata, each of the compressed outgoing digital voice data packets havingheaders and each of the compressed incoming digital voice data packetshaving headers; main control means connected to the telephone lineinterface means, connected for receiving the compressed outgoing digitalvoice data packets from the digital signal processor means, connectedfor receiving outgoing computer digital data packets from the personalcomputer through the communications interface means, operable formultiplexing the compressed outgoing digital voice data packets and thecomputer digital data packets to produce multiplexed outgoing data andfor sending the multiplexed outgoing data to the telephone lineinterface means for transmission over the telephone line: and the maincontrol means further operable for receiving multiplexed incoming datafrom the telephone line interface means, the multiplexed incoming datacontaining incoming computer digital data packets multiplexed with thecompressed incoming digital voice data packets, the main control meansfurther operable for demultiplexing the incoming computer digital datapackets and the compressed incoming digital voice data packets, and forsending the incoming computer digital data packets to the personalcomputer through the communications interface means and for sending thecompressed incoming digital voice data packets to the digital signalprocessor means.
 2. The module according to claim 1 wherein the digitalsignal processor means further includes acoustic echo cancellation meansfor removing at least a portion of the incoming digital voice data fromthe outgoing digital voice data for acoustic echo cancellation.
 3. Themodule according to claim 1 wherein the digital signal processor meansfurther includes line echo cancellation means for removing at least aportion of the outgoing digital voice data from the incoming digitalvoice data for line echo cancellation.
 4. The module according to claim1 wherein the digital signal processor means is further operable forreceiving the outgoing digital voice data from the full-duplexconversion means, for receiving incoming voice signals from thetelephone line, for digitizing the incoming voice signals to produce theincoming digital voice data, for removing at least a portion of theoutgoing digital voice data from the incoming digital voice data forline echo cancellation, for converting the outgoing digital voice databack into analog voice signals for transmission in an analog fashionover the telephone line.
 5. The module according to claim 4 furtherincluding a microphone and a speaker connected to the voice interfacemeans and wherein the digital signal processor means further includesacoustic path echo cancellation means for removing at least a portion ofthe outgoing digital voice data from the incoming digital voice data foracoustic path echo cancellation.
 6. The module according to claim 4further including digital gain means included in the digital signalprocessor for providing user-selectable gain or automatic gain controlto the incoming digital voice data.
 7. The module according to claim 1wherein the digital signal processor metres is further operable forcompressing the outgoing digital voice data into compressed outgoingdigital voice data packets by performing the steps of:a.) removing anyDC bias in the outgoing digital voice data to produce a normalizedoutgoing digital voice signal; b.) pre-emphasizing the normalizedoutgoing digital voice signal to produce a preemphasized outgoingdigital voice signal; c.) dividing the pre-emphasized outgoing digitalvoice signal into segments to produce a current segment and a pastsegment; d.) predicting the pitch of the current speech segment to forma pitch prediction; c.) calculating the gain of the pitch of the currentspeech segment to form a prediction gain; e.) reconstructing the pastspeech segment from a compressed past segment to produce a reconstructedpast segment; f.) finding the innovation in the current speech segmentby comparing the pitch prediction to the reconstructed past segment toproduce an error signal; g.) determining the maximan amplitude in thecurrent speech segment; h.) quantizing the error signal using a codebook generated from a representative set of speakers and environments toproduce a minimum member squared error matching the form of an indexinto the code book; and i.) recording the pitch prediction, theprediction gain, the maximum amplitude and the index into the code bookin a packet as the compressed outgoing digital voice data.
 8. The moduleaccording to claim 1 wherein the digital signal processor means isfurther operable for detecting silent periods in the outgoing digitalvoice data and for producing in response thereto a silence flag andwherein the main control means is further operable for transmittingoutgoing computer digital data packets on the telephone line when thesilence flag indicates the absence of voice information and wherein themain control means is further operable for multiplexing, modulating andtransmitting both the compressed outgoing digital voice data packets andthe outgoing computer digital data action the telephone line when thesilence flag indicates the presence of voice information.
 9. The moduleaccording to claim 1 wherein the telephone line includes a cellulartelephone link and wherein the main control means is further operablefor periodically transmitting a cellular supervisory packet on thetelephone line and for maintaining the link to the remote site over thetelephone line if the remote site acknowledges receipt of the cellularsupervisory packet within a predetermined period of time and fordropping the link to the remote site over the telephone line if theremote site fails to acknowledges receipt of the cellular supervisorypacket within a predetermined period of time.
 10. The module accordingto claim 1 wherein the main control means is further operable forreceiving fax data from the personal computer through the communicationsinterface means and for transmitting the fax data over the telephoneline.
 11. The system according to claim 1 wherein the telephone line isan analog telephone line and the telephone line interface memos isfurther operable for modulating the multiplexed outgoing digital datainto a modulated analog signal for full duplex digital communication.12. The system according to claim 1 wherein the telephone line is adigital telephone line and the telephone line interface means is furtheroperable for sending the multiplexed outgoing digital data over thetelephone line as unmodulated digital data for full duplex digitalcommunication.